[Asterisk-Users] Asterisk Dry Run

[niko singh]

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

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

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