Displaying 20 results from an estimated 500 matches similar to: "regarding sip.conf and extensions.conf"
2011 Feb 02
5
Regarding asterisk
Hi every one,
I am using asterisk version 1.6.2...... i did not
install mysql data base and when i tried to register a client from SIPp xml
file..... it is registered....
My questions are 1. where can i find that registered client?
2. when i type the command "core stop now" it exists and the registered
users are not shown why this is happening?
3. Is it compulsary
2011 Jan 25
2
regarding quit, exit and stop now in asterisk
Hi all,
i am running asterisk by using command asterisk -r, asterisk -vc
............ when i want to come out of asterisk it not getting exit or quit
from the shell....
i have tried soo many options like...
stop now
stop gracefully
exit
stop
quit
its not working stillll can any one tell me what would be the problem with
this?
please help me ... :(
with regards,
viswavardhan
2011 Jan 26
5
Regarding error in Asterisk dail plan:
Hi all,
i am doing my master thesis on server perfromance evaluation i am
using asterisk as sip proxy server and sipp tool as traffic generator...
i have run basic testing of asterisk like as shown in website
http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp
when i have copied sip.conf and extensions.conf from the site and run the
client and server i
2011 Feb 01
0
regarding error in asterisk
Hi all,
when i was trying to register a sipp client by using
register_client.xml file with .csv file in asterisk server i have
encountered an error that
1064 2150.240891 127.0.0.1 127.0.0.1 SIP Status: 481 Call
leg/transaction does not exist ....
I dont kknow how does this error comes and i have searched whole the
internet i am unable to find a solution for this.....
2013 Mar 09
1
[LLVMdev] Code morphing pass
Hello,
for an university course I am working on a code morphing pass. The idea
is that a random vector gets generated at the entry point of functions
and that vector will be used to randomize the flow of execution.
Alternative flows are built looking at the instructions inside the basic
blocks of the function and for some of them replacing the orginal
instruction with a set of logically
2015 Jul 20
1
apcsmart: doesn't detect missing battery on Smart-UPS 2200 RM
Michal,
I ran across this Debian bug today:
https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=707223
It's a bit of a corner case, and it's for NUT 2.6.4 (apcsmart driver v3.04), but I was wondering if it was still applicable for the latest code?
"nut reports 100% battery charge after unplugging the battery on a Smart-UPS 2200 RM (SUA2200RMI2U) using apcsmart.
Powerchute correctly
2013 Mar 09
0
[LLVMdev] Code morphing pass.
Hello,
for an university course I am working on a code morphing pass. The idea
is that a random vector gets generated at the entry point of functions
and that vector will be used to randomize the flow of execution.
Alternative flows are built looking at the instructions inside the basic
blocks of the function and for some of them replacing the orginal
instruction with a set of logically
2011 Mar 30
1
dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?
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
*>*
2005 Feb 07
3
SIPP load testing - unexpected message - anyone using sipp sucessfully ?
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
2013 Jun 20
0
Customer src in CDR with incoming sipp calls
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 ?
2013 Aug 27
1
Introducing Sippy Cup: SIPp Load Testing Made Easy
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
2013 May 20
1
Stress testing Asterisk
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
2007 Jan 23
2
stress-test realtime voicemail with sipp
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()
2010 Mar 15
1
Article - a method on how to evaluate an Asterisk server
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)
2012 Jan 11
1
Problems faced in load testing of asterisk
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:
2009 Apr 02
1
Trying to test my voicemail
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
2018 Mar 06
2
[OT] Load testing with SIPp
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
2014 Mar 11
1
PJSIP - Using multiple AOR contacts when dialing through an endpoint
Hello everyone,
I have started testing the PJSIP stack.
I saw that it is possible to setup statically multiple AOR contacts, setup
qualify_timeout and attach it to an endpoint, and then dial using this
endpoint.
When I setup the configuration I used the cli in order to see the status of
the contacts, and it worked fine - whenever a contact is unreachable, the
status is updated to Unavailable.
2004 Jun 01
0
Réf.: RE: SIPP Load testing
You maybe have to create a SIP user called like it is declared in your
UAC/UAS xml file. I think it should be 'sipp' or something like that...
-----asterisk-users-admin@lists.digium.com a ?crit : -----
Pour: <asterisk-users@lists.digium.com>
De: "C. Johnson" <javadude@cedrick.net>
Envoy? par: asterisk-users-admin@lists.digium.com
Date: 31-05-2004 08:03
Objet: RE:
2005 Jun 28
4
Anyone using SipP to produce RTP load?
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