Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

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.

Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable On Wed, Jul 1, 2020 at 10:32 AM Stefan Viljoen <mailto:viljo...@verishare.co.za> wrote: Hi Joshua HTTP is used on in our setup on http://127

Re: [asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

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

[asterisk-users] 13.22.0 - HTTP session count exceeded 100 sessions - instance unusable

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

[asterisk-users] Asterisk 13.22.0 unstable on Azure Centos 7 & cannot encode .gsm files

Hi all, I'm running a Centos 7 instance in Azure with Asterisk 13. The Centos VM has 24GB of RAM and identifies the CPU as Intel(R) Xeon(R) CPU E5-2660 0 @ 2.20GHz. This is a virtual copy of a physical Centos 7 machine which has 16GB of RAM and the physical CPU identifies itself as the

Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

...@verishare.co.za; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 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 <mailto:vi

[asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls

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: WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length

[asterisk-users] What are "non critical" invites?

Hi All I'm getting tens of thousands of these messages ever hour in the Asterisk CLI for Asterisk 13.22.0: [Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on 192420-502043860-301870737 on non-critical invite transaction. [Apr 20 15:59:46] WARNING[45462]:

Re: [asterisk-users] Predictive call - agent talking to a customer, then suddenly talking to another customer

Not sure if that helps much. Thanks for the reply! > 13. 2. 2020 v 19:06, Stefan Viljoen : > >  > 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 &g

[asterisk-users] Predictive call - agent talking to a customer, then suddenly talking to another customer

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

Re: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

Hi Patrick Wow ok thanks. I was not aware of this. As far as I can determine the Azure VM we're using was set up on the "stock" Centos 7 option Azure offers. So you're correct, it then won't be an official Centos 7 image. The one we are running on our bare-metal hosts IS installed from an

[asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

Hi all I maintain the above - it was set up by an external party with whom relations have now been severed by my employer. Quite early after the deployment it became evident that all .gsm audio files produced on this virtual instance at Azure via MixMonitor are corrupt. If you play back the

Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)

Hi Mike Oh yes, this: --- Note that the call ID is much longer than in the column display, e. g. a visual call ID in the sip show channelstats display may be 31f867c50ce but the full call ID is then 32d867a55cfb563b7f59da01de84d...@xxx.xxx.xxx.xxx:5060 The full ID can be obtained by typing

Re: [asterisk-users] Odd one-way audio problem (Mike Diehl

Hi Mike No rtp.conf in /etc/asterisk?! This might be part of your problem, AFAIK rtp.conf is part of any standard installation might it be that you have a corrupted Asterisk install and / or -other- missing conf files as well? Might this not also be part of your problem? I installed

Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)

Hi Mike In rtp.conf, what are the port ranges you specify? I had almost exactly the same problem not too long ago. People will phone, and sometimes it will work, sometimes not - one way audio would happen, then start working, then stop working. The problem turned out to be that the port

Re: [asterisk-users] Asterisk users survey

Hi Joshua Does the survey imply that there are big changes coming for Asterisk? E. g. features or facilities will be dropped / deprecated from the open source version in new releases, big changes to existing facilities / protocols, what is supported officialy by Digium for the official

Re: [asterisk-users] Problem with the DB() function (Ira)

>So the new install is coming along. I hooked up the new box for a couple of >hours and got a bunch more problems worked out. And yet some still remain. I >have this subroutine I call occasionally: >. >. >. >Also, when I installed asterisk it did not set itself up to start when the >machine

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)

work as far as "our" Asterisk was concerned. "Our" Asterisk could not distinguish if it was getting a call from a phone, or from this ViciDial server. Hope some of this helps. Regards Stefan -Original Message- From: Stefan Viljoen Sent: Thursday, 28 February 2019 08:18

[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM (Thomas Peters)

Hi Thomas What is your IVR box's domain name in Linux? With a hostname of, for example, "mcts.org" do you have a line like this in /etc/hosts: 127.0.0.1 mcts.org in your /etc/hosts? Additionally, in /etc/asterisk/asterisk.conf, is there a line systemname = that is -uncommented- and

Re: [asterisk-users] trying to upgrade asterisk and Debian -- not working

/configure, make delete all modules, followed by make install. On Thu, 24 Jan 2019 01:17:32 -0500, Stefan Viljoen wrote: > > What procedure did you follow to revert back to the old version? -- _ -- Bandwidth and Col

Re: [asterisk-users] trying to upgrade asterisk and Debian -- not working (John Covici)

What procedure did you follow to revert back to the old version? It sounds like your binary has been revereted, but the modules it needs to load are still the 13.24.0-rc1 modules... --- Hi. I am trying to upgrade my asterisk from 13.15 to the latest of asterisk 13 which seems to be

Re: [asterisk-users] Cannot originate to extension unless /etc/hosts is edited constantly? [Tony Mountfield]

s /etc/hosts is edited constantly? Message-ID: In article <018201d4acef$898a4b10$9c9ee130$@verishare.co.za>, Stefan Viljoen wrote: > Hi Guys > > I've run into a weird problem on Asterisk 13. Again something that worked > fine on 1.8 but is now broken on Asterisk 13. >

Re: [asterisk-users] Various extensions ring once and go to voicemail

Subject: Re: [asterisk-users] Various extensions ring once and goto voicemail - Thomas Peters >Carlos and Stefan (and other who have helped): >I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling >Asterisk is unrealistic in my position but I wonder if I can

[asterisk-users] Cannot originate to extension unless /etc/hosts is edited constantly?

Hi Guys I've run into a weird problem on Asterisk 13. Again something that worked fine on 1.8 but is now broken on Asterisk 13. I have an extension 3015. I'm trying to originate a recording playback call on it via AMI by sending Action: Originate ActionID: test Channel: SIP/3015 Exten:

Re: [asterisk-users] Various extensions ring once and go to voicemail - Thomas Peters

Here’s what I get: apbx*CLI> module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_dahdi.soDAHDI Timing Interface 4 2 modules

Re: [asterisk-users] Outbound caller ID ignored

Hi Mitch I don't think you can expect it to work on the analog lines at all out of the box. What kind of hardware are you using to connect to the analog lines? We used Xorcom Astribanks at a stage, but could never (ever) figure out the signalling needed to set the caller ID at our local

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joel)

>Hi, >On the other side.. There is a specific note regarding CDR behavior changes >from asterisk 12 onwards. So going from 1.8 to 13 means it applies to you. >Have a look at: >https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 >And

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR

>> Is it possible that the setup part of the call (between initiation and >> answer) >> is recorded in a separate CDR? >An excellent question. Unlike in the past versions calls can actually generate >multiple CDRs because CDRs now represent the flow of communication between >things.

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Tony)

In article <009d01d4a7e0$af2e0a50$0d8a1ef0$@verishare.co.za>, Stefan Viljoen wrote: >> 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

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)

Message: 2 Date: Mon, 07 Jan 2019 06:07:54 -0500 From: "Joshua C. Colp" >> On Mon, Jan 7, 2019, at 3:04 AM, Stefan Viljoen wrote: >> Hi guys . . . >> E. g. on 13, I see this (zero ringtime) for a call that I make to my >> cellphone to test, with my cellpho

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)

nowhere use the Answer() application in my dialplan when dialing out? -Original Message- From: Stefan Viljoen Sent: Tuesday, 08 January 2019 08:49 To: 'asterisk-users@lists.digium.com' Subject: RE: Re: Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp) Message

[asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero

Hi guys A few months ago I upgraded most of my Asterisk servers to 13 from 1.8. I've still got about 25% of my servers on 1.8. I've since noticed that ringtime on Asterisk 13 - the time difference between "start" and "answer" in the CDR record for any call, and between "duration" and

[asterisk-users] CURL to post application/json (David P)

>We tried to use the CURL fn to POST json, but it's sent as form data and >there seems no support for changing the Content-Type header. We switched to >invoking curl in the shell. Hi David If you've got if fixed that way, great. Just thought I'd comment and share the scripts / manner we use

[asterisk-users] Spontaneous reboot due to MySQL lookups ? (Jonas Kellens)

Hi Jonas We have more or less this behaviour with 1.8.32.3 when writing CDRs to ODBC on Percona 5.6 (MySQL drop-in replacement with some optimisations and extra features.) In our case the system does NOT reboot, but Asterisk itself crashes with a segfault inside unixODBC itself. We're

Re: [asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings - solved

Hi Guys Found the solution for this...! https://wiki.asterisk.org/wiki/display/AST/New+in+12#Newin12-channels_chan_a gent and https://reviewboard.asterisk.org/r/2657/diff/1/ and https://blogs.asterisk.org/2016/02/10/converting-from-chan_agent-to-app_agen t_pool/ clarifies the situation.

[asterisk-users] Asterisk 13.22.0 - No channel type registered for 'Agent' when queue rings

Hi All With the below config, I just keep gettings this in the Asterisk 13.22.0 CLI: WARNING[15872][C-0051]: channel.c:6343 ast_request: No channel type registered for 'Agent' whenever a caller gets sent to that agent queue with logged in agents waiting for calls on Asterisk 13. 3997 and

Re: [asterisk-users] SHELL() function Asterisk 13 - can only accept one paramter in string?

that come from outside, for example via a SIP header. Regards. -- Ludovic Gasc (GMLudo) Le ven. 27 juil. 2018 à 09:37, Stefan Viljoen mailto:viljo...@verishare.co.za> > a écrit : Hi all This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash sc

Re: [asterisk-users] asterisk-users Digest, Vol 167, Issue 17

ers] SHELL() function Asterisk 13 - can only accept one paramter in string? Message-ID: <008fecea-e2c0-9230-7cb4-cd1b6990c...@tootai.net> Content-Type: text/plain; charset=utf-8; format=flowed Le 27/07/2018 à 09:36, Stefan Viljoen a écrit : > Hi all > > This is a followup on

Re: [asterisk-users] Asterisk 13 - system() dialplan app cannot call bash scripts

Hi Guys Just feedback on this particular thread, this issue is SOLVED. The reason why SYSTEM() and SHELL() was not working for me was that I was passing a linefeed character (\n, hex 0x0a) in one of the channel variables in Asterisk that was then parsed in the call to SYSTEM() and SHELL(). It

Re: [asterisk-users] SHELL() function Asterisk 13 - can only accept one paramter in string?

With thanks to Ludovic Gasc --- Turns out there is nothing wrong with the SYSTEM() or SHELL() dialplan functions in Asterisk 13.22.0. After several hours of painstaking debugging, the problem turned out to be a linefeed (\n character, 0x0a hex) in the first parameter passed from Asterisk to

[asterisk-users] SHELL() function Asterisk 13 - can only accept one paramter in string?

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

[asterisk-users] Edit: Asterisk 13.22.0 - "stat" dialplan function clears channel vars?

Sorry, I see I have submitted a testing version of the dialplan fragment. Actual extension is: ;listen to recording exten=>,1,Answer() exten=>,n,NoOp(Requesting File ${recfile}) exten=>,n,NoOp(Rec file set to ${recfile}) exten=>,n,NoOp(Alt file set to ${altfile})

[asterisk-users] Asterisk 13.22.0 - "stat" dialplan function clears channel vars?

Hi Guys I have the following dialplan code that I use to play back recordings, the filename being provided in an originate statement to the AMI (AJAM) interface: Action: Originate ActionID: test Channel: SIP/3015 Exten: Context: local Priority: 1 CallerID: 3015 Account: recordinglisten

[asterisk-users] Recompiling Ast results in a binary with differing SHA256 sums?

Hi Guys If I recompile Asterisk (on a Centos 7 test box, Asterisk 1.8.32.3) multiple times in a row, e. g. make clean;configure;make menuselect;make I note that the asterisk binary in the /main folder in the source tree, has a different SHA256 hash each time I recompile Asterisk using the

[asterisk-users] Big leap - 1.8 to 15.4.0

Hi all We have to upgrade soon from prehistoric Asterisk 1.8.32.0 to Asterisk 15.x.x (whatever minversion is current at the time.) We are quite heavily invested into 1.8.32.0 at about 17 sites locally and internationally and have a LOT of custom software running our sites via AMI from various

Re: [asterisk-users] Passing parameter to Queue-called macro

ild channels with inheritance). Global variables are defined in a [globals] section in extensions.conf. (https://wiki.asterisk.org/wiki/display/AST/Global+Variables+Basics) -- BR, marie On 11.05.2018, at 9:01, Stefan Viljoen <viljo...@verishare.co.za> wrote: > Hi Marie > > Thanks! > >

Re: [asterisk-users] Passing parameter to Queue-called macro

)=1); return; } -- marie On 08.05.2018, at 16:16, Stefan Viljoen <viljo...@verishare.co.za> wrote: > 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 a

[asterisk-users] Passing parameter to Queue-called macro

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

Re: [asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

dphone issue. test on another phone / network plug. On Feb 9, 2018 2:48 PM, "Stefan Viljoen" <viljo...@verishare.co.za <mailto:viljo...@verishare.co.za> > wrote: Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who a

[asterisk-users] Call picked up from queue and transferred gets disconnected - about 0.01% of calls

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

[asterisk-users] Random, uncommanded blind transfers

Hi Guys I've got a situation where an incoming call originated from a trunk provider will ring in a call center and be answered by an agent. Usually within 15 seconds of the incoming call being answered, it will randomly blind-transfer to another extension in the same call center. It

[asterisk-users] .gsm recordings corrupted when running Asterisk in virtual machine

Hi guys Has anybody encountered a situation where .gsm recordings are corrupted if the Asterisk instance is running in a virtual machine? Specifically running Asterisk in MS Azure cloud VMs on Centos 7 inside the hosted instance. Switching to .wav files solves the problem, I can only guess that

[asterisk-users] CDR_TDS driver disappears - "does not provide a license key" on reload attempt

Hi All I have an Asterisk 1.8.32.3 instance that will at random intervals stop logging CDR data to MSSQL via FreeTDS. On investigation I'll find that the FreeTDS module has been unloaded somehow. It is not listed in cdr show status or show module like. Trying module load cdr_tds results in

[asterisk-users] asterisk.conf ignored?

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-users] Asterisk sip_autodestruct messages - extensions locked

Hi guys Does anybody have any opinion on what causes tens of thousands of these messages per hour to pop up in the CLI: [Jun 30 14:24:59] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct: Autodestruct on dialog '7e9597ae6ce95fef23374f4b380a9b70@192.168.0.1:5060' with owner

Re: [asterisk-users] ODBC locks warning in CLI - Asterisk 1.8.32.3

Hi all Just to report back if it is of interest to anybody, I managed to solve this. Had nothing to do with Asterisk, rather with Percona 5.6 / MySQL which is the database used via ODBC from Asterisk to store CELs and CDRs. I added the following in the /etc/my.cnf file for the Percona instance

[asterisk-users] Subject: Re: ODBC locks warning in CLI - Asterisk

further fixes" over a year ago. >11.x went into "security fix only" last month - 13 and 14 are the current versions - can you try with them? On 23 November 2016 at 12:52, Stefan Viljoen <viljo...@verishare.co.za> wrote: > Hi all > > I get this warning in the Asteri

[asterisk-users] ODBC locks warning in CLI - Asterisk 1.8.32.3

Hi all I get this warning in the Asterisk CLI about once every ten minutes or so: [Nov 23 14:47:36] WARNING[2544]: res_odbc.c:647 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: HY000: [MySQL][ODBC 5.1 Driver][mysqld-5.1.73]Deadlock found when trying to get lock; try restarting

Re: [asterisk-users] Asterisk 11.24.1 garbled audio

Date: Tue, 15 Nov 2016 17:52:07 +0100 From: Olivier To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 11.24.1 garbled audio Message-ID:

Re: [asterisk-users] 1.8.32.3 - billsec field does not not increment after call answer - what triggers it? (Joshua Colp)

is fine - it is just that Asterisk never "realises", as regards the CDR, that the call was in fact answered. At least it is working and audio flows back and forth. I'll see if I can come up with a SIP trace. Thank you! --- Stefan Viljoen wrote: > > Only this one trunk consistenly has

[asterisk-users] 1.8.32.3 - billsec field does not increment after call answer - what triggers it?

Hi Guys I've got a strange problem - on my asterisk instance, when a call starts to ring, I do core show channel and I get the usual output with the duration and billsec fields included. For most of my calls, things are normal, e. g. duration field starts incrementing as the SIP phone

[asterisk-users] Proper way to start Asterisk on CentOS 7? (Carlos Chavez)

Hi Carlos I have experienced something similar starting Asterisk 1.8.32.3 on Centos 7 on commodity / whitebox hardware. The problem was that Asterisk was starting "too quickly" in the systemd startup sequence, before the required services it needs to run were up and ready. I eventually came up

Re: [asterisk-users] Best timing source? (Carlos Chavez)

Hi Carlos We had similar issues with Asterisk 1.8.11.0 and Asterisk 1.8.32.2 with timerfd and pthread timing. The server instance would run for 20 or 30 minutes and then producde legions of error lines about timerfd or pthread, whichever we were using. The symptoms were voice quality issues -

Re: [asterisk-users] CEL entries over ODBC several hours late (Vinicius Fontes)

Hi Vinicius Thanks for replying. >Sorry for the probably obvious question, but it's better to cover all bases. >The DBMS is running on the same box as Asterisk is? If that's the case then maybe the DBMS is using too much CPU and starving Asterisk? I don't think so - I think I have a locking

Re: [asterisk-users] CEL entries over ODBC several hours late (Matthew Jordan)

Hi Matthew Thank you very much for the reply. I must have something seriously wrong somewhere else then - I retested now and the "apparent" effect is as I describe but your info definitely contradicts that. But you're obviously correct. One more question - I've noted that if I run a

[asterisk-users] CEL entries over ODBC several hours late

Hi guys I'm running 1.8.32.3 with CEL logging over ODBC to MariaDB 5.5.41 on the same Centos 7 machine. I've noticed that the CDR entries made are all in-time, e. g. the call will take place and the CDR entry is immediately written into the CDR table in the MariaDB database. However, CEL events

[asterisk-users] 1.8.32.3 - no timing indicated, tens of thousands of __sip_autodestruct error messages

Hi list Just to let everybody know I think I've got to the bottom of the above problem / error. Turns out that the issues described in my previous post were caused by problems in an MSSQL database that the Asterisk 1.8.32.3 instance was writing to...! Once I dropped the FreeTDS driver (while

[asterisk-users] 1.8.32.3 - no timing indicated, tens of thousands of __sip_autodestruct error messages

Hi all I'm running a 1.8.32.3 instance of Asterisk on Centos 7. About 80 SIP phones connected. I'm getting tens of thousands of error messages in the CLI, and callers are having extreme difficulty dialing, they can dial once and then if they try to dial again their extension is shown as busy.

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk (Stefan Viljoen)

Hi List Regarding this Asterisk instance as discussed previously (Asterisk 1.8.11.0) that was consuming enormous amounts of file descriptors (100 000+ for about 50 simultaneous calls) it appears I have managed to solve my problem by upgrading the 1.8.11.0 Asterisk instance to an 1.8.32.3 Asterisk

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

Hi D'Arcy that the server IP for RTP as specified in the initial SIP is correct? Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. This was a problem we had when the RTP server negotiated in SIP with our VOIP ITSP on one side of the

[asterisk-users] One way audio - doesn't seem to be NAT issue

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

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk

: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keepcomplaining Message-ID: assp.06651285b6.mqd94l$qb1$1...@softins.softins.co.uk In article 002b01d0d414$36af31b0$a40d9510$@verishare.co.za, Stefan Viljoen viljo...@verishare.co.za wrote: Anybody else ran into this? No, but I

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk

hardnofile 50 root softnofile 50 did the trick. We also tried vi /etc/sysctl.conf fs.file-max = 50 not sure what the solution in the end was. But I remember rebooting was important. Markus Am 11.08.2015 um 11:00 schrieb Stefan Viljoen: Anybody else

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

Hi Steve Just running about 50 calls?? If I do lsof | grep asterisk | wc -l to narrow the realm of what is reported I still get just under 100 000 files: lsof | grep asterisk | wc -l 95903 with 50 calls running. So apparently this is excessive? I can only guess that there must be a MAJOR

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

Anybody else ran into this? No, but I would ask myself why so many file descriptors are being used. It sounds like you have a file descriptor leak (not being closed when finished with). Hi Tony Thanks for replying. I suspected something like that, though repeatedly running lsof | wc -l

[asterisk-users] 786 000 files limit Centos 7 - Asterisk keeps complaining

Hi Guys I keep getting messages in the Asterisk 1.8.11.0 CLI that there are not enough file descriptors available on my Centos 7 box. I also get regular error messages that RTP connections are failing due to bad file descriptors. I have already edited /etc/sysctl.conf by setting fs.file-max to

[asterisk-users] CEL eventtime incorrect, but CDR times are correct - 1.8.11.0

Hi list I have a huge problem with a 1.8.11.0 Asterisk instance not logging CEL events with the correct eventtimes. I'm logging via ODBC to MariaDB 15.1 Distrib 10.0.20-MariaDB I'm logging into a MyISAM table. If I start the Asterisk instance, logged times are correct, but the longer the box

[asterisk-users] Filters

Hi list I'm using Asterisk 1.8.11.0 - is there any way to apply (for example) a bandpass filter to Asterisk RTP audio in the realtime audio stream? I'm looking for a way to (for example) filter out a 50Hz AC hum present in some calls I push through my asterisk. Thanks Stefan --

Re: [asterisk-users] Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?

Hi JG Thanks for replying. Depending on the hardware you are using, simply calling asterisk might not be enough, as there could be dependencies on third party drivers. Depending on how asterisk was installed, one probably also has to look at various permissions. For example, asterisk -r might

[asterisk-users] Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?

Hi list I'm trying to get Asterisk 1.8.11.0 to start automatically when my Centos 6.5 box boots. I've done this many times before, but for some reason, on this box and hardware (older Core i3 system, 4GB RAM) I cannot get Asterisk to be contactible after boot. E. g. in rc.local I have, as the

[asterisk-users] CEL eventtime drift

Hi list I'm running MariaDB with Asterisk 1.8.11.0 over ODBC for CEL and CDR logging. CDRs work fine, times logged in the DB are correct consistently. However, I've noticed that for CELs eventtimes start lagging severely. E. g. I'd start Asterisk at 12:15 and entries made into CEL and CDR

[asterisk-users] Asterisk 13.4.0 - mixmonitor only records one side's perspective

Hi All I have a problem with mixmonitor in 13.4.0 doing the following: 1. Caller phones in 2. Reception picks up 3. Talks to caller 4. Does attended transfer, talks to manager to screen the caller wanting to speak to him 5. Complete the transfer by putting down her handset so the caller can

[asterisk-users] For a failed retransmission - what were the IP addresses?

Hi Guys Given these occassional errors on my Asterisk CLI: [Jul 2 10:23:36] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 17bb3a993ad10f8818970ae952b81e73@192.168.11.31:5060 for seqno 102 (Critical Request) -- See

Re: [asterisk-users] Strange and complete failure of Asterisk 1.8

Re: Strange and complete failure of Asterisk 1.8 (Duncan Turnbull) Re: Strange and complete failure of Asterisk 1.8 (Markus Weiler) Thanks Marcus Duncan Pulled the machine and replaced it with a brand new one. Same network and same DNS server active there. New system is running the same

[asterisk-users] FW: Strange and complete failure of Asterisk 1.8 - part 2

Hi guys I just did a ps -Af | grep asterisk on the machine and got several screens full of this: root 6970 6946 0 13:10 ?00:00:00 rasterisk rxcore show channels verbose root 6987 6948 0 13:10 ?00:00:00 rasterisk rxcore show channels verbose root 7005 6985 0

[asterisk-users] Strange and complete failure of Asterisk 1.8

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

[asterisk-users] Too many open files - 786 000 already specified as max num open files?

Hi guys I have a site on Asterisk 1.8.11.0 running in Centos 6.5 that has about 150 concurrent callers. I keep getting these types of messages in the CLI: [May 21 11:39:21] WARNING[18469]: channel.c:1189 __ast_channel_alloc_ap: Channel allocation failed: Can't create alert pipe! Try increasing

Re: [asterisk-users] Monitoring SIP Service (Jai Rangi)

Interesting approach. What we've done is to write an app that runs on a separate machine that simply does some asterisk -rx calls to the running Asterisk instance via an SSH library and then evaluate the string returned. For example, to monitor our registered SIP service providers, we compare

Re: [asterisk-users] asterisk-users Digest, Vol 130, Issue 14

- Original Message - From: Steve Davies davies...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 13, 2015 11:39:29 AM Subject: Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32

[asterisk-users] Linking Asterisk 1.8 to late model Samsung PABX over PRI - transfer issues

Hi all I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk 1.8.11.0. The reason is the customer has a large installed base of Samsung phones physically connected to it and on each users desk. They

Re: [asterisk-users] System() command refuses to execute bash script

Hi all I got this solved. Turns out the script WAS executing, but I forgot that apparently you need to follow cron rules in any BASH scripts executed via System() from an Asterisk dialplan. E. g. all paths must be fully and absolutely specified, there are no relative path references available.

[asterisk-users] System() command refuses to execute bash script

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

Re: [asterisk-users] TimerFD errors if MTU size is set incorrectly - SIP trunk

Hi Guys Regarding this I found the following links which appear relevant: https://issues.asterisk.org/jira/browse/ASTERISK-19347 https://issues.asterisk.org/jira/browse/ASTERISK-18223 It seems that this issue is related to that, NOT to a too-large MTU size. Don't know if anybody can comment?

[asterisk-users] Timer_fd, pthreads, or DAHDI timer for timing under 1.8.11.0?

Hi guys I have some questions regarding the above 1. Why are there different options for timing? 2. What are the differences between these types of timing sources? 3. When should you use what? 4. Is one timer type more reliable for an Asterisk system under heavy loads than another while NOT

[asterisk-users] TimerFD errors if MTU size is set incorrectly - SIP trunk

Hi all Is there a relation between the above? I'm having a problem where I suspect my internet access provider (through whom I go to a SIP trunk provider) have got MTU size problems. My asterisk (1.8.11.0) is constantly going into the situation where a TimerFD error is spammed in the CLI, load

[asterisk-users] 1.8.11.0 - CLI error res_timing_timerfd

Hi all Sometimes (about every three months) some of my Asterisk 1.8 boxes will start running this message thousands of times in the CLI: [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]:

Re: [asterisk-users] constantly increasing load in Asterisk 11.14 (Sebastian Damm)

Have you considered doing a daily reboot? In our shop (about 14 sites, busiest doing about 90 000 calls per day) we found it best to reboot each Asterisk instance at 23:45 - we're still running an ancient version (1.8.0.11). Once this regime was instituted our major Asterisk issues stopped,

[asterisk-users] Meaning of core show hint output

Hi all If I have the following in my dialplan: exten=25001,hint,SIP/25001 Doing a core show hint 25001 results in 25001@local : SIP/25001 State:IdleWatchers 0 1 hint matching extension 25001 in the Asterisk CLI. What does the Watchers 0

Re: [asterisk-users] Asterisk executable suddenly about 40KB

What you may want to consider is if you have a network management system such as Nagios is create a service that checks the size of the binary every 5 minutes. You're notified if the size goes over a certain threshold. You can also take the perf data and graph it using one of the many Nagios

Re: [asterisk-users] Asterisk executable suddenly about 40KB larger - modules (Andres)

I would also start by putting an audit rule on the binary. Something like this: auditctl -w /usr/sbin/asterisk -p war -k asterisk-bin then you can get a report on who modified it and when by using: ausearch -f /usr/sbin/asterisk Its a start, but eventually you might need to monitor even

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