similar to: Which CDR processing for high load ?

Hi, I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I don`t mean ?I don`t like it?. I mean it crashes the server. I realize there are multiple CDRs per queue call ? one per ring/per phone, basically. The issue is that whenever the number of CDRs ?to be recorded? for a call exceeds 5000,

Hello, SIPp's PCAP play feature can replay pre-recorded audio stream towards destination (see [1]). Doc mentions tcpdump and Wireshark as tools to record such RTP streams without further details. Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/ directory. Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to 10.1.6.18:2006 1. How can you "forge" IPs

Hello, How would you test how a PJSIP-powered Asterisk 13 instance resist to hostile REGISTRATION attempts ? Would you use SIPp ? Any example scenario ? Would you go with an alternative tool ? Which one would you pick ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

Hi, Reading this old thread, may I ask if keeping hangup handlers from updating CDR values still enforced in Asterisk 15 ? If positive, would it be very complex to add in Asterisk, a configuration option allowing a system administrator to list in cdr.conf, the CDR fields allowed to be updated in hangup handlers ? I'm planning to store some RTCP stats. Saving them in CDR(userfield) would be

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

Thanks for the feedback. I do agree with having multiple smaller servers. When I was first approached with this task I mentioned as much. However, the current desire is to work with already existing hardware. That is out of my hands at the moment unless it just can't be done. I will explore Freeswitch a bit soon to compare it as well. I am struggling to find what the bottle neck is in

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()

Hi everybody, got it from svn: dtmf_2833_1.pcap */asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN *>*

Hi list, I added custom code to eliminate frame index references. I replace each FI reference with a subtraction from my frame pointer register (just like ebp in x86). Its result is stored in another register which is used by the load/store instructions. Nevertheless, this operation gives me redundant subs that I would like to remove after all fi references have been eliminated. For example:

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The

Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729

I have data that plots over time with four different variables. I would like to combine them in one plot using facet_grid, where each variable gets its own sub-plot. The following code resembles my data require(ggplot2) require(reshape2) subm <- melt(economics, id='date', c('psavert','uempmed','unemploy')) mcsm <- melt(data.frame(date=economics$date,

Hi, RFC3680 defines a SIP event package for registration. This event package which can be used through NOTIFY-SUBSCRIBE methods, seems very useful for free sitting or presence applications. This package is supported in various SIP phones (at least Thomson ST2030) : when turned on, this feature adds a new login/logout menu among other things. It can also be used to send Welcome notices to mobile

Hello, I'm running load testing sessions. My System Under Test is an asterisk 13 with 16GB, configured with maxfiles set to 400 000. This system is supposed do produce simple SIP trunking services without transcoding. The box sending call to my System Under Test is anabled with SIPp. I'm banging on a 700 concurrent calls/50 CAPS limit I would like to improve, if possible. Tests are

Hi, I just installed Sipp 3.3?on CentOS 6.3 and all of the calls Sipp is generating are failing. I am trying to run Sipp on the same machine as Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. SIpp output: ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- ? Start Time???????????? | 2013-05-20?22:53:08:637?1369083188.637273??????????? ? Last Reset

Hi friends... I am trying to test my voicemail with Asterisk using SIPP (SIPP is running in Debian Squezze and Asterisk is running in OpenSuSE-11.1), the command that I use is: sipp -sn uac_pcap -l 1 -m 1 -s 55 -trace_err 192.168.13.6 But, If I use the file g711a.pcap included in the sources of sipp or if use some file captured for me the result is the same ---> error ... the message in

Hello all, I would like to share with you an article [1] we have issued last week (sorry, currently only in Romanian language - we plan to provide an English version soon). This article is describing a method to be used for obtaining the maximum number of SIP simultaneous calls an Asterisk server could process safely (meaning no errors/maintain control of the machine and without RTP frame drops)