[asterisk-users] Beginning to use Asterisk and tests with extensions

[Daniel Bareiro]

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> Hello Daniel,

Hi Dana.

> You will find the information at http://www.voip-info.org/ and 
> http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the 
> "Online Book" link) very useful.

I have the second edition that covers Asterisk 1.4 and it seems
interesting. You made me remember that I had downloaded it the last
year, although just now I have more time to dedicate to Asterisk. The
fact of to have already installed it is an important step :-)

> The asterisk package by itself should be adequate for SIP/IAX calls.
> I don't think you need libpri unless you are planning on connecting asterisk 
> to a digital connection such as ISDN or a PRI.
> You will need Zaptel (for Asterisk versions 1.2,1.4) or DAHDI (Asterisk 
> versions >=1.6) if you choose to install an internal card (OpenVOX, Digium, 
> Sangoma, etc.)      I do not know if or how well this will work with a VM.

Thanks for the indication. According to I saw in the site of
Asterisk[1], only make reference to DAHDI for Asterisk 1.4, but
according to which you say to me, both can be used.

My idea is to buy an ATA to connect a conventional telephone and make
tests of communication between it and softphone. The idea by which I
thought about using an ATA is because I am not sure with my version of
KVM (KVM-62) can make PCI pass through. But with the ATA must not have
problem.

Having this in mind, I installed the packages dahdi-linux-2.1.0.4.tar.gz
and dahdi-tools-2.1.0.2.tar.gz having loaded only the module dahdi_dummy
and so far commenting all that appear in /etc/dahdi/modules.

> I suggest testing your SIP softphone with the Echo() and/or Playback() 
> dialplan applications before attempting to call another 
> softphone/hardphone/etc.   This will allow you to confirm that the one 
> endpoint functions properly before adding more complexity by calling another 
> endpoint.

I was testing and sometimes with Echo() and MusicOnHold the sound is
broken. Is there some form to solve this?

> some things that allow you to call a conventional telephone:
>     an ATA with an FXS port
>     an internal card (such as OpenVOX, Digium, Sangoma) with an FXS port
>     call a conventional phone number through the PSTN (below)
>
> To connect to the PSTN you can use any of:
>     an ATA with an FXO port (plug an analog phone line into it)
>     internal card with an FXO port (also to plug an analog phone line in)
>     account with an ITSP (there is occasionally discussion on the list about 
> advantages/issues/opinions/and flames with various ITSPs - google 
> "site:lists.digium.com ITSP")

> [...]

I believe that with the example I understood a little better how it
works. As it mentioned above, I am thinking about buying a Linksys
SPA3102 to make both internals and with PSTN tests.

> Hope that gets you going in the right direction.
>
> http://www.voipsupply.com/ is a good source to see what equipment is 
> generally available to end users. 

Thanks for your reply and by all the references and examples that you
provided to me.

Regards,
Daniel

[1] http://www.asterisk.org/downloads

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