Setting up outbound calling.

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Setting up outbound calling.

Postby musicace024 » Sun Apr 01, 2012 3:50 am

Hi everyone,

I installed the dialer and everything works fine... up until trying to set up the phone system. I'm really confused on how to set up a phone line/sip/voip with it. I'm using version 2.3.0. I don't know how to accurately add a carrier, link it to an agent or extension... or anything. I have an account set up with freecalls.com, I also have a magic jack, google voice number and a cell phone. Volunteers will be calling from home to help, so if I could set it up to use cell phones, that would be perfect! If someone has a chance to help me, I'd really appreciate it!!

Best wishes,
Ben
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Re: Setting up outbound calling.

Postby fadmin » Mon Apr 02, 2012 9:56 am

This technology is not a Home solution, but requires a bit more robust environment. Cell phones, Google Voice, Magic Jack, freecalls.com, etc. are all the type of things which is not compatible with this type of a dialer. For one these type of carriers will all close your account for violating their acceptable use policies, and / or not workable, the moment they realize that you are using a dialer.

OSDial requires either a SIP carrier or T1/PRI service from the phone company. They provide the information needed to configure OSDial over SIP. Based on what you say it sounds like OSDial is not the type of solution that is intended for your type of operation. I'm predicting lots of technical issues as well as security issues where someone will take control of the dialer and rack up huge bills for you to pay.

I am trying to scare you away, not because I dislike you at all, but because of recognizing a type of confusion and total lack of understanding of rudementary basics which most of the time will result in disasters. Much like an unhappy marriage I'd rather not get married than go through all the unhappyness of an incompatible match. (Are you familiar with all the the laws governing call centers?) I'm suspecting that you need to do a lot more home work. Have someone with experience give you a hand and explain what works and does not. There are lots of pitfalls which runs away with costs.
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Re: Setting up outbound calling.

Postby musicace024 » Tue Apr 03, 2012 11:36 pm

Admin,

I appreciate your help, thank you.

I understand that you're trying to scare me away from doing this, and I appreciate you bringing the potential fears to my attention. The last thing I need is a high phone bill! I admit I'm not very well versed on the server side of professional telecommunications. However, I work for a call center and I've been constantly researching the legalities of professional fundraising for the past year. What you saw in this post was a very small bit of the homework I've been conducting. In short: I know a lot about operations and call center legalities, but my technical knowledge is a LITTLE weak (pathetic) as you've seen!

If you know somewhere I could read about setting up securities and setting up a virtual call center, I'd really appreciate it. And a reliable (but cost efficient) SIP carrier would be nice, too!

Best wishes and thanks again!
Ben
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Re: Setting up outbound calling.

Postby musicace024 » Wed Apr 04, 2012 1:44 am

UPDATE: Okay, I just found an SIP service (voipvoip). Now when I make an outbound call, it dials and then when the "donor" (me on a cell phone to test) takes the call, it disconnects immediately and tells me that I was hung up on. What do I do?
musicace024
 
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Re: Setting up outbound calling.

Postby fadmin » Wed Apr 04, 2012 10:46 am

Sounds like mismatched codecs or a firewall issue. With SIP it starts on port 5060 but once the connection is established the voice part goes on a random port (usually) in the range of 10,000-20,000. Which means you better allow the ports your carrier is using.
At the same time you should only allow it from your carriers IP, or others are likely to take advantage of it.

Another note, Asterisk does not handle the high traffic of a dialer without having an external timing source, typically a USB stick from Sangoma called UT50 / UT51 (internal/external). In a virtual environment lag is introduced which prohibits Asterisk from accessing it in a timely fasion. The end results are missed hangups, cross-connects and other "interesting" effects.

Another well priced SIP provider is WSS Communications (http://wsscommunications.com). Their prices are much better and they can provide fiber to the premise. Let them know we told you about them and they will take good care of you.
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