Could anyone give me any pointers as to how to make outgoing calls from trixbox ce 2.8?

guya
Posts: 9
Member Since:
2009-10-20

The company I work for are looking to replace their legacy telephony applications (a mixture of analogue/digital cards), various pieces of third party software etc, and consolidate all of the products using trixbox/Asterisk.

I have created a trixbox 2.8.0.1 virtual machine and installed freetds 0.82, pymssql 1.0.2 and Python 2.6.3. I've managed to write a number of python scripts to enable incoming calls to be taken and processed.

I have created a free account with VOIPtalk and so we have 1 incoming number. I've also added some credit to the account in readiness for the next stage.

What I now need to do is find out how to make outgoing calls.

I've searched these forums and all relevant places on the Internet, but none of the resources seem to provide an "idiot's guide" (which is what I need).

I'm not at all familiar with the terminology (trunks, sip etc).

I'm hoping someone here can answer a couple of questions/issues...

1. In order to make outgoing calls, will we need any additional hardware fitted to the trixbox server (I realise that I'm using a virtual server at the moment, but I can build a dedicated physical one if required). We don't want to hook up any other VoIP equipment to this server (phones etc). It will simply receive calls, update any number of SQL Server databases and also be capable of making outgoing calls as and when required.

2. At the very least, would we need a sound card installed in the trixbox server?

3. The telephone numbers that would need to be dialled could not be "hard-coded" into trixbox. They would be derived from a SQL Server database.

4. Is this something that would be possible using Python/AGI?

5. I connected to the CLI console to try and find out if there were any dial commands and there are, but only if you enable something called "chan_oss" in the modules.conf file (apparently). I tried this, but I still couldn't use the "dial" command. Again, I'm sure I'm missing something completely obvious.

I think the biggest problem I have is that there are so many technologies involved (Asterisk, Python, freetds, trixbox), and that, combined with all of the terminology is making it difficult for me knowing what to search on to retrieve the most useful information.

If anyone could even give me a few pointers as to where to start, I'd be very grateful!

Regards



nttranbao
Posts: 189
Member Since:
2008-02-16
Google for "Trixbox without

Google for "Trixbox without Tears".

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IT/VOIP consultancy, VOIP eStore, Support Forum
Bao Nguyen IT Co., Ltd.
http://www.baonguyen.vn
WE MAKE IT



SkykingOH
Posts: 9678
Member Since:
2007-12-17
You also would be served

You also would be served well to hire a consultant to at least frame requirements and guide you to what interfaces to use.

The normal way to make calls with the trixbox distribution is from a phone to a trunk (either a pots line or Voip). In your instance their are numerous API's that can connect calls.

You need to understand Asterisk channels and API's

I would run to www.asteriskdocs.org and read the book cover to cover. Buy a hardbound copy and support the project!

--

Scott

aka "Skyking"



vccs
Posts: 19
Member Since:
2008-09-04
I agree, a consultant would

I agree, a consultant would be a great idea. Python scripts should not even be needed becasue everything is already built into the software.
No hardware is necessary with a VoIP infrastructure.
SQL is not needed to store much information unless you're using asterisk Realtime.

If you are looking for initial or ongoing support and good termination please get a hold of me :)

Thanks.

--

Matthew Martin
Chief Consultant
VCCS
http://www.vccs.ca
trixbox Hosting, Customization and Support|Business VoIP



guya
Posts: 9
Member Since:
2009-10-20
Many thanks for all the

Many thanks for all the replies.

I had glanced through "Trixbox without Tears" a while ago, but it seems to concentrate more on IP phones and devices.

Our VoIP provider (VOIPtalk) has several links on their web site to give advice on how to set up a trunk for outgoing and incoming calls, which I've followed.

I did create a ".call" file and drop it into the /var/spool/asterisk/outgoing/ folder and I then see the call trying to be made in the Asterisk Console.

Typically, the output I see is..
Attempting call on IAX2/VOIPtalk/xxxxxxxxx for s@my-outgoing:1 (Retry 1)
Hungup 'IAX2/VOIPtalk-15890

the content of the ".call" file is
Channel: IAX2/VOIPtalk/xxxxxxxxx
MaxRetries: 5
RetryTime: 10
WaitTime: 45
Context: innovise-outgoing
Extension: s
Priority: 1

One thing I have noticed is that in the PBX Status in the IAX2 Registry section, all I keep seeing at the end of the line is "Request Sent" followed by "TimeOut". Is this significant for making outgoing calls.

Again, many thanks for everyone's help so far.



guya
Posts: 9
Member Since:
2009-10-20
Apologies for the double

Apologies for the double post, but I've just dropped another ".call" file into the outgoing folder and my mobile phone has just rung as a result of the call!

No idea what I changed, but I guess at least I've proved the concept.

My final question is "Is using .call files a bad thing?". What I mean is, is it bad to construct .call files and move them into the outgoing folder as a way of making calls?



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