similar to: How to size an email server to handle 5 million emails per day

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition?

Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, today a customer asked how to use asterisk with 12 to 16 FXO ports. I can use a channel bank with 16 FXO ports and connect the channel bank with a T1 cable to a T1 card in the Asterisk Server. Asterisk will then send the calls to the Voip provider over the internet. However a 16 fxo port channel bank is about USD 1500 + a t1 card USD 500 + a USD 1000 computer = 3 thousand us dollars + my

Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, One customer has Centos 5 installed. He wants to have a web based frontend to the vacation program because he is in charge to enable the vacation msg for all the users in leave. I tried webmin but the webmin vacation module points to a nonexistant link. And the usermin module is very old and requires the user to do it by themselves. Suggestions? --

Hi people, Do you have pointers to web documents that help me make comparisons between buying a server with two quad core 2.33 ghz or buying a 4 dual core 2ghz server? I am trying to answer a question of performance. It is not important the redundancy/failover or the price of the server. Just the performance. obviously all the hardware specs are the same, the question is the CPU. --

Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten => 12345678,1,Answer() exten => 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -----Original Message----- From: Erick Perez [mailto:eaperezh at gmail.com] Sent: Thursday, July 26, 2007 7:03 AM To:

Hi, I use putty in my windows xp machine. ssh server in a centos 4. The centos 4 machine runs a web server that listens on port 1812, the centos machine is behind a firewall that allows tcp 22 connections only. I am on public internet. Can I forward/redirect/allow my web browser in windows to "see" the web page in port 1812 of the centos machine via the SSH connection? Thanks, --

Hi, im using 0.99 stable in RPM form, Im using this for postfix+dovecot+real unix users, now i tried to enable virtual support with this: protocols = imap pop3 ssl_disable = yes log_path = /var/log/dovecot.log info_log_path = /var/log/dovecot.info login_user = dovecot mail_extra_groups = mail auth = default auth_mechanisms = plain auth_passdb = pam auth_userdb = passwd auth_userdb =

I have this server I need to put to work. The option I have is to make it work as a small office PBX with SIP users and a Digium E1 Card for PSTN service. 24 SIP users and one E1 card in an Intel 945board (533 Front side bus) with 1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB 533MHZ) and two 80GB SATA disks. Can the box sustain the load? I can add another 1gb of ram if necessary.

the user you are connecting as should have full rights to /var/run/asterisk: http://www.voip-info.org/wiki-Asterisk+non-root hth -----Original Message----- From: Erick Perez [mailto:eaperezh@gmail.com] Sent: Monday, April 03, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk.ctl

Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big

Benson, no SELINUX was not enabled. The instance was selected without it just to make things easier. I do not have a pull request for the installation manual yet. On Tue, May 5, 2020 at 1:21 AM Benson Muite <benson_muite at emailplus.org> wrote: > > On Mon, May 4, 2020, at 10:38 PM, Erick Perez - Quadrian Enterprises wrote: > > Hi Centos friends. > > I had some time to

Hi All I have been able to get MP 104 FXO to make outbound calls with my asterisk box and polycom IP 500 phone. However I cannot get the incoming calls to hit the asterisk box. Any help will be appreciated. Thanks, Lal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060612/36de60cc/attachment.htm

Hi, haven't found anything in google's, i wonder if there is a comparative page of what to expect from running * on motherboards like the EPIA and similar ones. Since i have not used *ever* such kind of mini atx form factor boards, I have no clue about their performance. SIP-SIP communications, voicemail SIP-TDM communications, voicemail how may users (SIP hardphones and analog phones

Hi, I am about to give a training session in Linux, using Centos 4.4 as a base. I want to give my students (MS windows administrators) printed books about Linux and hopefully Centos based. It is a total of 20 books. I'm also interested in purchasing 20 copies of the DVD of Centos 4.4 x86/x86_64 distro. My initial search at amazon returned no useful results. It can be RHEL 4, Centos 4 or

Just a question, has anyone knows how to blank or factory reset an AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit). I purchased them second-handed with no manuals (thank god for the internet!!) but i guess the pdf manual I have does not have the section of factory-reset. Also, any sucess stories with: AudioCodes MP124-C/FSX/AC/SIP

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

Hi, Im using dovecot version 0.99.11 and I'm facing a situation with one pop3 client. This is a fax application (faxmaker for windows) written by a third party that tries to login to the dovecot as USER username at domain. When the application tries to login that way, my dovecot rejects the auth. All the other pop3 clients (outlook, eudora,etc) in the system log in correctly because they do