similar to: different _source_ addresses for registrations?

Displaying 20 results from an estimated 800 matches similar to: "different _source_ addresses for registrations?"

2006 Feb 15
5
Aasterisk large-scale deployment w/analog phones
hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar
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
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()
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
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
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)
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
2005 Sep 23
2
ZAP ISDN losing digits
Hi all, I got into a strange problem here. I've got an asterisk box with bristuff-0.2.0-RC7k, and a HFC PCI ISDN card, running in NT mode. The ISDN card is connected to a S0 bus and to a Siemens ISDN PBX. Two phones are connected to the ISDN PBX and are successfully getting calls from the asterisk box. When dialling from one of the phones, the ZAP channel seems to be missing out on some of
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
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
2018 Feb 09
3
[OT] How to use audio files with SIPp
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
2015 Aug 19
3
asterisk server stress test
Hi Barry Flanagan, Barry Flanagan <barryf-lists at flanagan.ie> schrieb am Mit, 19. Aug 11:06: > SIPP is probably what you seek. http://sipp.sourceforge.net/ > > Hope this helps. That looks pretty like what I'm looking for! Many thanks! Sincerely, Dominique Haeber
2008 Apr 22
2
Asterisk sends 486 Busy Here instead of 600 Busy Everywhere
Hi, We have a scenario wherein the endpoint needs to send a 600 Busy Everywhere after receiving an INVITE. I am using SIPp as this end point. SIPp is configured as UE2. Now when UE1 calls UE2 (SIPp) receives the INVITE and responds with a 600 Busy Everywhere. But when Asterisk receives this 600 response it sends out a 486 Busy Here to UE1. Ideally Asterisk should be relaying the 600
2011 Apr 13
1
Asterisk thread limit
Hi Guys! I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could handle in production so following is my senario. [sipp_client]---------------[Asterisk]----------------[sipp_server] sipp_client ./sipp -sf uac_pcap.xml -d 100000 -i 172.30.254.211 -s 2000 172.30.1.47 -l 1000 -r 250 -rp 5000 -m 1000 sipp_server ./sipp -sn uas -i 172.30.245.208 In above if i set -r
2015 Nov 06
2
bad performance centos6 ->centos7
hi, i'm evaluating performance of centos7 i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough - sipp generators/asterisk receiver with answer/playback) scenario - sipp generators - asterisk - asterisk receiver (i wrote ansible scenario for this if you are interested) then i reinstalled system to centos7 x86_64/distro
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:
2007 May 28
2
help on asterisk sipp
Good morningI was wondering whether you could help me. I installed sipp on my Asterisk server but I don't really understand how does it fonction! Has someone ever tried it?If you can explain to me the principle, I would be extremely grateful.Thank you very much in advance. _________________________________________________________________ Lancez des recherches en toute s?curit? depuis
2010 Jul 12
1
[LLVMdev] [PATCH] Start of SIMD Reorg
Bruno Cardoso Lopes <bruno.cardoso at gmail.com> writes: >> Ok to commit? > > I'm Ok with this patch. > > Despite that, I think we should discuss the ones to come, If you really go > "tablegen auto generates everything" as I've noticed in some tablegen patches > you commited, there's a great chance the sse/avx code would become unreadable, >
2005 Sep 05
2
DTMF issue on IVR
Hi All, I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt and asks caller to dial four-digit extension. Caller has to dial slowly, otherwise, Asterisk cannot recognize the extension number. I look at the trace on Asterisk CLI and there are missing digit in the middle of string. ex, caller