[Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN

[Steve]

I'm going to try and ask this again and keep it short and as too the 
point as I can while still providing enough info to be of use.
PLEASE advise if I am going about this wrong or asking too much.
I'm seriously doing my BEST to throughly read the docs and try a bunch of 
things BEFORE coming here to ask and possibly annoy.
If is documentation that explains thsi process in terms that someone new 
can grab onto quickly I'm missing it!

OK here goes again :-)


I have complied an asterisk system and got it going from scratch and all 
works great except I cannot make an outbound sip-to-PSTN call and do not 
fully understand how to configure it.

I've been folowing some examples and keep running into this stumbling 
block:

As soon as I add (to sip.conf) this section:

[siprovider.com]
type=peer
host=sipprovider.com
fromuser=2135551212
secret=2135551212
authname=2135551212
fromdomain=siprovider.com


I no longer can recieve ANY inbound calls from the PSTN via my sip 
provider.

I've tried many variations of attempting to get this section (I think it's 
referred to a 'sip channel) into my sip.conf all which give the same 
result.....

All inbound calls from PSTN TO this account FAIL.

I have tried with the dialplan in context [default]
with a test dialplan and with a 'blank' dial plan.

every way I try this, inbound calls via SIP and my SIP provider stop 
reaching my asterisk box.

If I remove the above shown section leaving only the
register => 2135551212:2135551212 at sipproviderexample.com

all works great and calls come in from the PSTN to my asterisk box and 
people can get around my menu just fine and dial internal SIP extension 
numbers.

This of course leaves me with no SIP [brackted] section of which to use 
for outbound calls of which I'd love to eventually get working.

Am I doing this right at all???? or am I headed completely in the wrong 
direction here?

I also tried this with a FREE Stanaphone account and get a very similar 
but strange result....

IN both cases adding this section to sip.conf result in my calls 
terminating at the SIP provider voicemail system instead of coming into my 
asterisk box here.



A side note:
Not that it really matters but here's what I get from my provider if I try 
to dial into my PSTN number from the PSTN:

Standard unavailable voicemail message as if not registered in.

-and-
>From Stanaphone:

a 'strange' voice message that gives you the option to either #1 change my 
outgoing unavailable message or #2 press ANY key besides #1 to hangup.

Any help or pointers would be GREATLY appreciated!!!!!


I compiled and Installed Asterisk about 10 days or so ago and am running
version:
CVS-Nv1-0-7-05/19/05-11:22:20 built by root at Vontage on a i686 
running Linux

Thanks!!!

Having a BLAST!


Steve Gladden