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21

Yes. Being in a position currently where the powers that be bought a Panasonic PBX. There's a bunch of features that aren't there (or rather, are individually licensed). Such as MeetMe (conference calling), or hot-desking. There's also the problem that all the phones are effectively hard-wired to their ports. They had to run new cables for the phones, ...


19

In this day and age you cannot directly compare a soft switch to a PBX. A soft switch, or software that connects phone lines with one another, is at the core of all modern PBXs. Asterisk is not a PBX but it can be configured to function as one. Also most of the functions provided by an PBX can also typically be performed by a class 5 soft switch. Perhaps an ...


14

Generally speaking, the more extraneous software you run results in two problems: Reduced memory availability for the software you actually care about A greater likelihood that your server will be vulnerable due to an unpatched security vulnerability. If your application is not memory intensive, or under particular memory pressure for your hardware, you ...


9

Well, there's a whole bunch of things wrong with what you're trying to do. You can't edit a file with ./ That's linux slang for "execute this thing in my current path". .conf files aren't traditionally executable. To edit a file, use some text editor like gedit, emacs, vi(m) or nano. Try reading the manual pages online for asterisk, or try man sip.conf ...


9

A good rule of thumb comes from old telephony .. an phone T1 line 24 voice channels and a control channel. This equates to roughly 64Kb/s per phone line. This rule works (roughly) for uncompressed VoIP. There are various codecs that do compression .. the G729 should roughly triple that, so get 72 voice channels on a T1. If you expect all 55 phones to be ...


8

Spend the extra money. I'm running my second Asterisk box and wouldn't do it without echo cancellation. I know it's a painful cost but since we make hundreds of outbound calls a day I wanted to take every step possible to eliminate echo (ie we are the source of the echo). I did this even though the only analog line we use is for our fax machine. Most of ...


7

I have come across the same problem and it's not just limited to asterisk. In the end we came up with something that worked well for us. We called it a SIP Loopback. Basically we signed up for an ALT sip provider (flowroute.com) and setup a script that calls out via primary SIP provider to the phone number setup with our ALT provider three times an ...


7

ServerFault doesn't really do shopping advice - you need to consult with vendors and decide what system works best for you. Some points to consider: Do you want a fully VOIP system (IP within the office, your PBX connects via VOIP to a provider that connects you to the PSTN), or a hybrid system (IP within the office, and your PBX connects to the PSTN ...


6

My thinking is to provide the GUI for folks who may have to succeed me and who do not know Asterisk so well as to manually maintain the configuration files. This seems like a silly assumption from the start. For one thing, virtually all of the documentation available for asterisk is in the form of config files, someone approaching your setup from no ...


6

Upfront: I used to be a core dev on freepbx, but moved to another company, and unfortunately don't have time for it anymore. It really depends on your style. Certainly, by hand, you get a ton of control. However, you also have to consider what you are trying to accomplish. In the end, an office PBX generally needs to work in a certain way, and there a bunch ...


5

I saw that you tagged Cisco in our question, so I am guessing you are using Cisco phones, I have only used Cisco phones with Asterisk, so that is all I will be able to cover in my answer. The first thing you will need to do is add a directory entry to your SIPDefault.cnf file (inside the TFTP directory your phone points to. Here is the line you need to add: ...


5

It works but there are applications that depend on an accurate clock that need some special attention, for example conference bridging. A good test is to install asterisk + zaptel and compose a test dialplan for a conference bridge. Connect to the bridge with two phones and listen to the audio.


5

I believe I've figured it out. Since SIP clients and servers both operate on port 5060, I can't have an Asterisk server running on the same box as my softphone. I've installed Asterisk on a remote server, and I can connect to it with Ekiga with the instructions from the book just fine. Ha, the book even says the same. Shame on me. If you are running ...


5

In the past for similar testing situations, I've just made a T1 crossover cable. The TE405P is a quad-port card, so on the failover system, just connect two of the ports back-to-back with a T1 crossover, then write a quick script to have asterisk place a bunch of calls out one port, answering them on the other port.


5

That server will be a killer for Asterisk, you can expect over 5k+ concurrent calls without degradation, although on this scenario if you use some advanced codec (G729 for example) the CPU will be your limit. Have a look at this very good document about Asterisk dimensioning, lots of very useful info there ...


5

A timing source is exactly what it sounds like - a reliable source of a fixed frequency signal used for timing (Asterisk uses it for, among other things, the RTP stream and recording playback). Timing sources may be software, or hardware (via supported analog or digital line cards). Hardware timing sources have the advantage of not being affected by system ...


5

A PBX (Private Branch Exchange) is a device to switch calls between telephones. Some like Asterisk support VOIP (Voice Over IP). When I looked at implementing Asterisk a few years ago, I found I was looking at a hefty hardware expense for the phone interfaces. FreeSWITCH is a VOIP switch and handles switching calls between VIOP endpoints (connections). ...


4

In your output, there is an error from the format_wav.c module saying the input is not a wav file. Check your format and make sure that you're saving it in the format Asterisk expects. As a workaround, you can use the open source audio tool "sox" to make things a little easier: sox /var/lib/asterisk/sounds/custom/Untitled24.wav -r 8000 -c 1 ...


4

The system is waiting for further input to determine whether the complete extension is '1' or something else that starts with '1'. Do you have other extensions in that context that start with the number 1? Check your dialplan.


4

Your remote user will need to register their IP phone to your Asterisk-based system, which means you will need to either allow non-local addresses to register, or provide a VPN or similar tunnel so the IP phone behaves as if it is local. Once either of these are accomplished, you simply create a SIP extension and password on the Asterisk server, configure ...


4

I would separate your question into 2 areas: Going VoIP Going Open Source Regarding the first one, I believe there is not much dispute, being VoIP extremely more powerful than traditional PBX, but to name a few: Usage of same network infrastructure you already have laying around, reducing cost Ability to connect remote endpoints without expensive ...


4

We use a phone system called Swyx (which is similar in functionality to Asterisk, but runs on Windows). As a part of the call route scripts, we can execute external scripts. We just wrote a very small script on our shitlist extension that puts the phone number of the caller into a special table that gets scanned when a call comes in in our automatic call ...


4

This is how I secure my Asterisk server, which has been in production continuously since 2006. Firewall Open inbound ports only for necessary services. (You do have to open a wide range for RTP streams, but this generally isn't an issue since nothing normally listens within that port range.) 22/tcp ssh (for management, of course) 4520/udp DUNDi (if you ...


4

If the UPS units are drained, there's probably not much you can do without power. Your goal should be to make sure you have sufficient UPS/generator capacity for your PoE switches and your Asterisk device to be up for however long you need in a disaster scenario (20 minutes is fairly short, I'd be aiming for an hour). It wouldn't hurt to have some ...


4

A few general points -- Asterisk does very poorly in virtual machines. I don't know if they have specific guidance on this, but my experience is even in high-end VMWare environments you can get into all sorts of odd trouble with virtualized VoIP servers. I would strongly advise troubleshooting on dedicated physical hardware. SIP (the VoIP protocol behind ...


4

If you aren't comfortable with telecom systems and the idea of assembling your own system from the available (software and hardware) components on the market I would suggest a nice, packaged Asterisk solution, such as the Trixbox CE appliance or an appropriate model of PhoneBochs. You can get these with appropriate line cards (Analog or Digital/T1) and set ...


4

You can use a conditional expression to match against the MAC address of the phones: if substring(hardware, 1, 3) = 00:04:f2 { option routers 10.0.32.1; } else { option routers 10.0.32.5; } I tested this inside a subnet stanza, but I believe it will work anywhere that an option command will work. Tested with ISC DHCPD 3.1.2.


4

Your initial impressions are correct; the GUIs like do to things their way, and attempts to step outside of that box will sometimes be difficult. Think of the way that say Debian does their Apache configuration vs the standard source distribution - one config file vs tens or even hundreds being pulled in via include statements. You will also tend to find ...


4

If you have the bandwith (roughly 100 kbps per call), G711 should give you best call quality (and least CPU usage in case Asterisk is transcoding). Otherwise, G729 is a good choice. These two should be supported by most hard- and softphones. For more exotic codecs, check what your phones support.



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