similar to: asterisk on Vmware

Hi,. I have some legacy applications for MSDOS / 95 and a new NT version of an application that i would like to use. I have un-commented the lines in wine.conf about serial access but i cant seem to get any MSDOS or win32 application to talk, is this currently supported in (the lateset) wine ? regards --- Matthew J Fletcher NPD Firmware When did ignorance become a point of view ? ---

Hi, currently I''m using 48 class with htb & very stable Is there any maximum number of class I can create in a single linux box ? I need 500 or even 1000 class for campuss network. Any help appreciated thanks & regards Tino _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hi, I''m new to rspec and wanted to translate some of my unit tests into rspecs. Unfortunately my first test fails with "Mysql not loaded". My application is running fine and can access the database. I''d appreciate any help, so I can get started. Tino -------------- next part -------------- An HTML attachment was scrubbed... URL:

>Information on timing sources can be found here: >https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces >As noted on that page, ConfBridge can use any timing interface Asterisk >provides, and is not limited to the DAHDI timing interface. Generally, >timerfd is a good timing interface. >That aside, I would try to rule out external issues with the garbled audio

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, 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 am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

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

Hello, How to take the values of channel variables like 'agi_uniqueid' and 'agi_callerid' in agi script. For example #!/bin/bash -x T="$agi_uniqueid" I want to save value of 'agi_uniqueid' channel variable into a variable called 'T' in my script -------------- next part -------------- An HTML attachment was scrubbed... URL:

Winbindd can see the NT-user, but samba can't work with the NT-user. My System: SuSE Linux 7.2 Enterprise Server Samba-2.2.3a I have install samba by the following steps: 1. ./configure --prefix=/opt/samba-2.2.3a --with-winbind 2. make 3. make install 4. cp /tmp/samba-2.2.3a/source/nsswitch/libnss_winbind.so /lib 5. ln -s /lib/libnss_winbind.so /lib/libnss_winbind.so.2 6. vi

Hello, There is a xen setup in which "brctl show" gives the following output. bridge name bridge id STP enabled interfaces eth1 8000.003048c9d4df no peth1 vif1.0 vif2.0

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

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, I am quite new to R - but quite expierience in programming. Nonetheless I have some problemes in accessing information of the "summary" object. Here is what I do: model <- lm ( y ~ myVariable ) "summary(model)" gives me an object which has a lot of information about the regression. Now I'd like to access programmatically the level of significance which is

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)

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I