similar to: 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? Is the MTU size a red-herring as relates the timer-fd errors? Thank you very much Stefan

Hello All, I'm sure this has been talked about and based on some searching of archives, I'd discovered that to be able to use timerfd, one needs to have a kernel version >=2.6.27? Is this true? If yes, then is there anyone who's got it working in Lenny 5.0.7? Do I need to download and build the linux kernel (currently at 2.6.37) from scratch and get access to the TimerFD source?

Hi All Just finished setting up a vm with centos 5.5 and asterisk 1.8.3 Using timerfd as a timing source. Has anyone got a similar setup in production ? How's performance? Thanks, Neeraj?

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 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 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 using any DAHDI hardware - SIP only? I'm of course referring to the familiar

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 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

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, 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 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

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 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

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 queuing to Local/xxxxxxxxxx at local-0002dbea;2 WARNING[71993][C-00022dcc] channel.c: Exceptionally

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 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 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.

I am currently having a voice quality problem with one of our Asterisk servers. We have checked the network and we have found no problems that could cause the voice to sound cracked and with small interruptions. I am looking at the timing source for Asterisk and it is currently using timerfd even though we have an E1 card installed. Is timerfd better than dahdi? Any recommendations to

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

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