similar to: Asterisk Capacity

Hello, I'm looking for some advice on securing Asterisk. Recently my servers been under several brute-force SIP attacks. I have several remote sites, as well as many roaming users, who may have PC softclients and/or SIP based hardphones. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hello, I have a question about best practice (or recommended practice) for allowing SIP registrations from the Internet. This is what I was thinking of implementing: 1. Use OpenSips for the SBC, enable SRTP and TLS 2. Allow limited access to the actual Asterisk PBX (behind firewall) via OpenSips Is there anything that I am missing that probably should be implemented? Thanks, Richard

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

Somebody knows if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [*40 at liberado15:15] Record("SIP/1201-083453c8", "/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3") in new stack ??? -- <SIP/1201-083453c8> Playing 'beep'

Hi, I know somebody is going to give me the link to the wiki hardware pages, but I can't find the answer there. I'd like to know if, for an Asterisk only system (nothing else of note running on it), I get a real gain from having 2 CPUs. Does the amount of traffic/SIP registrations/codec translation possible doubles with 2 CPUs? (each quad core E5420 to be precise)? Does it

Right, I'm baffled. I have: exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2) exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\ (caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\ VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12)) exten => s,n,MYSQL(Query RESULT1 ${DB1} SELECT\

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Thanks for your reply. We are trying to implement a simple Mips-based CPU with just for teaching purpose, so we delete some instructions which are not commonly used, thus the task won't be too hard for students. I am responsible for modifying the compiler so that the compiler won't emit unsupported instructions. In order to avoid "can not select" error, I am trying to expand

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

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I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)

Hey All, Growing call center. Currently at about 200 call center staff, running about 1000 calls per hour. Gearing up to double that. Not too sure that a single server will support that growth. So, I'm trying to come up with ways to scale the system and still maintain a simplistic design. So I'd like to bounce some ideas around. Currently I am running on a Dell 1950, dual quad core

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Hi, We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. I've found an issue when Asterisk tries to make a SIP call out using auth, but has the wrong credentials and keeps getting returned a SIP 407, in this example to an OpenSIPs server requiring user auth. Basically this happens: 1. Asterisk sends plain INVITE to OpenSIPs 2. OpenSIPs responds with SIP 407 auth required with a

Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the