[Asterisk-Users] Asterisk Dry Run

[Greg Hill]

On Sat, 7 Aug 2004, niko singh wrote:

> I just installed asterisk on my system with the purpose of rerouting
> calls on sip channels. I don't think i need any hardware for that.

you're right, mostly. There are some asterisk features like meetme
conferences which require a timing source. This could be from a Digium
card or derived in software from some USB chipsets or from some RTC's. But
for basic sip channel stuff, you won't need additional hardware.

> I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both
> of them and to call one from the other on the same machine, however could
> not.
> I 1-) I could connect sjphone in isolation to freeworld dialup howver i got
> no sounds ...the records at free world dialup confirmed i made and recieved
> calls. My sound card works absolutely fine..however i got no echo or ring,
> yet the record shows i did make a call when i did.

not getting sound back through sjphone calling direct to fwd might be a
nat problem.. you didn't mention anything about the arrangement of your
network, though, so it's anybody's guess.

> 2-) With asterisk when i enter "sip show registry" it does not show either
> device...however they do appear in sip show peers at CLI. Are the devices
> registered or no.

"sip show registry" shows what sip servers you have registered asterisk
with. In other words, those servers would see asterisk as a client,
just as asterisk will see your softphones as clients. So this is the
expected behavior. (unless you set asterisk to register with FWD, in that
case it should have shown up in this list)

> Could someone kindly prescribe a few tests to check my asterisk
> installation ( i get the CLI) but now it seems may be sound devices are
> a problem...will installing an x100p solve it ?( i know it is an fxo but
> probably it uses its own on board sound config)

no, an x100p won't change anything. It's just a PSTN interface card.

> 3-)Are there any other free sip accounts i could check with..last night
> fwd seemed unreachable

it isn't sip, but you can try iaxtel. IAX is a protocol developed with
asterisk. It's similar to sip but has many improved features.

> 4-) I went to the LIPZ4 (zultys) config file in /usr/local/zultys/mp4_*.cfg
> and added my internet settings ...what changes do i need to make if i wish
> to use it to connect to fwd straight or connect to asterisk.
> 5-) I am runnign askterisk and all these on the same machine ..could someone
> kindly send me a sample zultys config. and some explanation.

I played with the lipz4 for a few minutes and couldn't get it working so I
dropped it. Maybe somebody else knows how to make it go.

> 6-) Absurd question perhaps but i was just thinking if this is a
> possibility : My number ( PSTN ) is 5678 ..now if someone dials 56789056
> will i get the call ? can i get 9056 as extension.

that would depend on your PSTN provider. But that's most likely not to
work. (your phone number is really only 4 digits long??) However, somebody
could dial 5678 and you could answer that and send it to an IVR (menu)
where they could dial the 9056 extension.

> Lots of questions...new here and would be grateful if someone from the
> communtiy could anser them

be sure to visit www.voip-info.org, www.asterisk.org, www.google.com, etc.

After spending time reading the above sites you'll be better equipped to
set up your system, as well as to know what information about your setup
is useful to us so that we can help with any problems you might encounter.

Greg