similar to: Asterisk Test Suite error

Thanks for replying. So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian setup the alternatives are either : - to install it from source - tto build my own custom package removing this asterisk dependency (is it easy or even possible ?) - to use another solution such as pyst. Regards 2014/1/16 Adolphe Cher-Aime <acheraime at gmail.com> > Yes you can. This what

Hello, Is it possible to run Starpy and Asterisk on different machines ? A quick glance at http://www.vrplumber.com/programming/starpy/ seems to tell it is possible but Debian's python-starpy package installs Asterisk. What do you think ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm trying to use SIPP (http://sipp.sourceforge.net/) to stress-test our asterisk installation. We have a very simple dialplan that uses FastAgi. I'm finding that all calls to "GET VARIABLE" from the FastAgi are returning null when the dialplan is invoked from sipp -- and they work fine when invoked from a softphone on the same machine, for example. Does anyone have

Hi there, I need a Python interface to asterisk manager for my own project. The voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+manager+Examples) lists 4 python projects for this purpose: Fats, py-Asterisk, pyst and StarPy. Because my project is rather small and I don't want to involve twisted in, the options left for me are py-Asterisk and pyst. So I want to ask your opinion:

Hi, I've installed a brand new Asterisk 12.0.0 system in which I can see, with make menuselect, in Test Modules tab, that each test entry such as test_acl can't be installed due a to missing TEST_FRAMEWORK(E) dependency. Where this TEST_FRAMEWORK(E) comes from ? How can it be installed ? I've installed testsuite according

I have built samba 3.0.8 with ADS support. From the Solaris 9 end it appeared to join the active directory server domain OK. However when I look using Windows Explorer on the ADS (Windows 2003) machine it appears as Samba 3.0.8 server under WORKGROUP, and I cannot access the shares. I am using the MIT kerberos. The Howto guide on page 74 is a bit ambiguous. I read it to say that if you are

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

Hello, I'm pleased to announce the first release of Obelus, a MIT-licensed Python library to interact with Asterisk using the AMI and AGI protocols. Compared to existing libraries, Obelus is framework- and programming-style-agnostic, and compatible with Python 3 as well as Python 2. It also has an integrated test suite. This is version 0.1, and as such some APIs are a bit draftish and not

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 *>*

I'm trying to test IMAP in 1.4.17 and it appears to be not working. I've compiled imap-2007 with the following on a CentOS 5 box: make slx EXTRACFLAGS="-I/usr/include/openssl -fPIC" and I've configured and compiled asterisk with the following: ./configure --with-imap=/usr/local/src/imap-2007 The compile and install went just fine, no warnings and no errors that I saw.

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

Hello, I'm stressing an Asterisk 11 platform with incoming calls from sipp 3.1. I've dedicated a context to sipp in my dialplan. Everything works OK expect that calls from sipp comes in with a CallerID set to sipp and this sipp value is stored in CDR. 1. I can change the value of the CallerID but how can I have the calls from sipp traced in CDR with a customized src field value ?

Hello everyone, Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives

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

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()

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)

Dear List after a lot googling and watching source example of FastAGI i cant find a simple way to convert a very simple perl AGI script... perhaps im not a developer.. Why i have need to use FastAGI?...Very load CPU usage on my box... with only 100 calls.. So i have two way res_perl or FastAGI on some other box.. I cant test res_perl becasue when i try to compile it i have this error:

Hello, I am trying to run load on asterisk server(version 1.8.7.1) through SIPp tool for the voicemail() application. But I am facing a lot of problems. I tried running 1000 calls?from SIPp for 100 subscribers (10 messages for each subscriber). I am using odbc storage for the messages. Following warnings/errors are coming on the asterisk server: Jan 11 11:30:49] WARNING[22924] app.c:

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, 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