I have been set out on a mission by the company I work for to install a in house pbx, and to create a predictive dialer that calls numbers from a .csv file. And shows the â€œlist infoâ€
I’m quite new to the asterisk community, but I’ve had a positive experience with polycom phones. Also, and this is just an outsiders observation, it seems ploycom is the phone manufacturer most asterisk administrators choose.
I’ve worked with Digium and Sangoma T1 hardware. I should mention my experience with this equipment is limited to a lab environment, not a production system. Both cards were easy easy to setup and there is plenty of online documentation available for both as well.
What is the best setup for this configuration hardware wise?
would a T1 be enough, if I choose to use sip for xlite on the computers, for a predictive dialer, and also do the in house pbx with asterisk.
And also does asterisk have issues with flash transferring, aka, not closing the channel, or dropping the customer from the transfer?
What’s ‘flash’? If you mean hook-flash in the analog world, the polycom and snoms have a transfer button. No ‘flash’ problem.
Asterisk is an excellent platform. You won’t lose channels or calls. Asterisk has two call legs. The internal phone to the PBX and the PBX to the T1. When you transfer, the T1 leg will be put on hold whilst you do what you have to do to get the next person on the phone. Call parking is a good habit to get into as well.
As for the predictive dialer. I have limited experience with real-world projects. I personally haven’t set it up for a couple of years. From my understanding it depends a lot on the length of calls, how good your list of numbers is etc…
What about the computer hardware, and t1/digum card hardware, whats the best to use for the current setup i need, based on what i have said earlier:
“”""I have been set out on a mission by the company I work for to install a in house pbx, and to create a predictive dialer that calls numbers from a .csv file. And shows the â€œlist infoâ€
Keep the fax out of it - it’s do-able, but to be honest, on your first implementation, just leave it out because it’s not part of the core requirements.
T1 is digital - ISDN.
If you have 11 analog lines, then you’d be looking at some kind of analog card. These used to be called TDM400 or TDM2400 cards.
If you have SIP clients on a computer, invest in USB headsets, not the shitty sound card types.
I was looking at the snom 300 phones, they look good, as for the t1, im on it. I can keep the fax out, how about the asterisk server, whats the best hardware configuration that would be in my best interest. also I was thinking of setting up a web server for our site, we have it on a hosted plan now, but would it be wise to do this in house with the asterisk server, or just keep it hosted, i want as much bandwidth to stay for the pbx/predictive dioaler system.
I’d get as much soup under the hood as you can afford. My experience with Asterisk is a simple setup like mine works on about anything. But your multi-user setup will require serious speed and as much memory as you can jamb in it.
Also, unless you’re desperate, keep your web server hosted elsewhere. First of all the hosting services these days are cheap, reliable and they take care of everything including backups, email, etc… for you. And anything besides Asterisk running on your Asterisk server will only slow things (your calls) down and make them sound jittery.
for the asterisk server, I chose this, let me know what u think:
Dell poweredge t300:
dual xeon 3.0 6mb cache 1333 fsb
3gb ddr2 667 mhz 6x512mb single ranked dimms
raid 1 config sata/sis
2 250 gb 7.2 k 3gbps hdd
intel pro 1000 pt 1 gbe dual port nic pcie-4
750va ups 120 volt stand alone
and a power connect 2716 web managed switch 16 port
as for the cards i need for the snom phones whats the best to get digium wise?
That is a lot of grunt.
9 computers needing sip clients for predictive dialer login and setup. (sip)
What is this requirement?
Looks like you’re going with handsets.
snom300 is a simple phone. perhaps you could order a snom300, snom320, polycom320 to test which you like - a phone system is an integral part of the business - if you get a handset that is not suited to your staff or needs, they will hate the system, blame it for anything that goes wrong and it will come back to bite you. Try all three, then sell the two you don’t want on ebay - total cost is a loss of 30% on two handsets - but you’re sure you have the better of the three handsets.
The phones talk to asterisk.
Asterisk talks to the card which goes out your T1.
The phones you choose have nothing to do with your T1.
If it’s all on one box 3GB RAM is good. If the DB will be on another server, then asterisk doesn’t need that much RAM to run.
Use a power over ethernet switch.
9 computers needing sip clients for predictive dialer login and setup. (sip)
This is where i need xlite on the computers using sip, to utilize the vicidial predictive dialer, the web interface it uses to handles the call list aka (.csv) file I will upload into the dialer, when the dialer calls the list gets an answer it will call the xlite sip account on the computers and the employee answers via usb headset, they also need the snom phones for hand dialing. completely separate from the handset/predictive dial system, so basically 9 computers will have xlite installed using a web interface predictive dialer via vicidial, and also have astrerisk running the snom/polycom phones. I want the employees to use vicidial to call the list i upload, and also be able to have the snom /polycom phones for hand dialing if needed becuase we use printed call list also.
whats getting me is that is it wise just to lay vici dial or gnu dial for that matter over asterisk and let them both run on the same server.
see basically the employee is sitting at the desk looking at the screen the web interface shows caller id it notifies the employee via tone a call is in from what the predicitive dialed from the list and the conversation starts .
When setting up this system do i need to setup a sip proxy server also?
No. Asterisk itself is a SIP Proxy. It acts as a good enough SIP proxy for most needs.
You will find contradictory comments and articles when dealing with much much larger scale installations.
Let me just clarify.
If you really want to know more about SIP, the first two chapters of Packt Publishing’s “Building Telephony Systems with OpenSER” is a good read. OpenSER is a SIP Proxy for high performance (10000’s of calls) where you would put it in front of other SIP telephony services (e.g. asterisks…)
Asterisk can act as a SIP Proxy.
You may want to, for your own sake, look up a few things to do with SIP telephony - because you will no doubt get a few recommendations. A little knowledge on the subject is a little dangerous. For instance, I have spoken to quite a few network engineers (and telco engineers) and just the way they talk differs depending on which books they read, which companies they work for, which courses they’ve attended, which questions and answers they’ve heard at whilst at seminars and ‘sewing circles’ around ‘the water cooler’. They are experts in their field, but telephony is a little bit different.
B2BUA - asterisk is also a B2BUA - back to back user agent. This basically means that asterisk will keep track of the state of a call on both sides of it - e.g. the handset of the user and the far end (usually a SIP registration to a phone company).
SBC - Session Border Controller - same as a B2BUA.
SIP Proxy - directs SIP messages, handles registrations etc…
Location Server - this is a part of a “SIP Server” that keeps registrations - a SIP proxy will receive a REGISTER message, and will pass it on to the SIP Registrar (again another little part of a ‘SIP Server’). The registration is kept in a Location Server (which is just a part of the SIP Server). This means that when a call request is made, the SIP Proxy asks itself “hey… do I know where I can send this phone number?” - if it can answer itself with “yeah, send it to XXXXXX@10.0.0.200” it will send the call there. If there is nothing in the location server, then it routes it to ‘the next hop’ so to speak - and this is where your routing rules come into play.
So it looks like you played around with asteriskNOW - if it has freePBX, you’ll see that you have inbound and outbound routes to configure - basicallly this is your sip routing - if a phone number is not in your internal config (extension registrations, ring groups, queues etc.) then asterisk will consult it’s routing rules you configure
If you’ve got the asteriskGUI, it’s got similar functions called “calling rules” from memory.
Hope this little bit of ‘dangerous information’ is useful enough for you to google a few things and be armed with a bit more knowledge.
Whats the best voip company to use for this, or are there any suggestion s on various ones. I ned something that will implement the sip trunk and dids and voip all in one and has a good standing and credentials.
or if anyone knows,
any other route I should take