similar to: Trying to test my voicemail

I am trying to use Asterisk as the voicemail system of the TELCO where I work. I wanna test with 20000 mail boxes ( and later with a better machine/server I hope try with 70000 ). How do I include in voicemail.conf the file with the mail boxes?, In a big system like this,is better use text files or any database? Thanks -- Linux User Registered #232544 Jabber : pepo at

Hi friends. I am working in a TELCO, we have a trouble with our very old Alcatel Voicemail system (and now we dont have support and worst this system was forgotten for Alcatel) I've used Asterisk for just small jobs, but I've proposed use it and tomorrow begins with the tests :) ... so: we have a thousands of users, how do I have to configure my server Asterisk to works like

Using two Asterisk connected between they, How do I can check the voicemail in a remote system but working like *97? I mean dont want ask the voicemail box, just the password and go to the voicemail of caller. If I have the same extensions in the two Asterisk it doesn't work. Thanks. -- Linux User Registered #232544 Jabber : pepo at jabberes.org Ekiga : pepo at

Hi friends... I am using pxelinux but it is so slow when I don't write MAC addresses in /etc/dhcp3/dhcpd.conf so How do I can use pxelinux.cfg/default directly and not have to wait? A lot of thanks. {pepo} -- Linux User Registered #232544 Jabber : pepo at jabberes.org ICQ : 337889406 GnuPG-key : www.keyserver.net ----------------

Hi friends. I am using Asterisk like voicemail of a great system with many users, How do I can get statistics of each box in the voicemail system? something like space, number of messages, etc. A lot of thanks. -- Linux User Registered #232544 Jabber : pepo at jabberes.org Ekiga : pepo at ekiga.net ICQ : 337889406 GnuPG-key : www.keyserver.net

Hi friends. Can you help me to use SIPP to stress my asterisk voicemail? I want to send my own recorded media file to the voicemail system. Thanks. -- Linux User Registered #232544 Jabber : pepo at jabberes.org Ekiga : pepo at ekiga.net GnuPG-key : www.keyserver.net ----------------------------------------------- dum loquimur, fugerit invida

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 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

In authlogic, I set the password field to "crypted password" Is there a way to display a password, even if its "crypted"? What if the user forgets the password and needs to recover it? How can I recover a crypted password? Thanks -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send

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)

Hello friends. I have a softswitch in which I cannot create a list of blocked source numbers; So, I have thought to use Asterisk and return a 302 message when the number can make the call, my dialplan is as follows: [from-external]   exten => _AX.,1,Verbose(=======> ${CALLERID(num)} to ${EXTEN})    same =>      n,Set(MYDESTINY=${REPLACE(${EXTEN},A,)})    same =>     

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

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, 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

Dear Samba Team ! Today i tried to install SAMBA4. In the last days / weeks I?ve made this very frequent and up to now it was working fine for me. I was testing SAMBA4Alpha15 with Debian Squezze Testing version. But since 2 Days I cant install SAMBA4 on new released Debian6 machines anymore. The command ./setup/provision --realm=samdom.example.com --domain=SAMDOM

Hello, I have asterisk 1.4.4 running with anonymous sip calls enabled and I am testing the box for load using sipp with something like this - sipp -sn uac -s 10 -d 60000 -i 192.168.1.49 -l 110 -r 5 -trace_err 192.168.1.50 Asterisk picks up the call and runs a test php-agi file that plays a .gsm file. As soon as the number of active calls reaches 99, asterisk starts Declining further calls. I

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

Package: xen-hypervisor-4.0-i386 Version: 4.0.1-2 Severity: normal File: xen -- System Information: Debian Release: 6.0.2 APT prefers proposed-updates APT policy: (500, 'proposed-updates'), (500, 'stable') Architecture: i386 (i686) Kernel: Linux 2.6.32-5-xen-686 (SMP w/2 CPU cores) Locale: LANG=en_US.UTF-8, LC_CTYPE=en_US.UTF-8 (charmap=ANSI_X3.4-1968) (ignored: LC_ALL set

I built asterisk-trunk at 92526 and asterisk-addons-trunk at 496. I have my Bluetooth cell phone connected. It almost works. In mobile.conf, I have "context=incoming-mobile" for the phone, and that looks like: context incoming-mobile { _. => { VoiceMail(9999,b); Hangup(); }; } Calls to the cell phone get directed answered by Asterisk and directed to

Hi all, I saw that the u32 selector can match a PATTERN so I can use it to filter on a port value like : tc filter add dev eth0 parent 10;10 protocol ip prio 100 u32 match ip sport 0x0014 0xffff flowid 10:100 (filter on the ftp-data (20) source port) But is there a method to filter all the source ports > 40000 for example ? regards, ------------------------------------------------ Franck