[Asterisk-Users] Please help I fear I have missed something very important! but what?

[Stuart Buchanan]

I had already tried that, however my register statement already specifies to
ring out on ext 1001 so the call isn't an unqualified one so should not look
up the s extension 

register => 2xxxx:xxxx at fwd.pulver.com/1001

However I put the s extension statements in and the results were the same
with or without the /1001 on the register statement.

When I run the SIP DEBUG command from the console, I can see the CLI of the
phone I am calling from in the sip statements however the asterisk box just
sends out an engaged tone.

A bit of background to the box, it is running a full install of Redhat 9
without the firewall enabled. (I did a full install as I thought it would be
good to learn about Linux as well)

It is not behind a NAT and configured with a static Public IP.

Does anyone know of problems that can occur with a full install of Redhat 9?
As I have read somewhere that someone recommends Redhat 8 and not 9 but he
wasn't specific as to why, does anyone know of any problems (or is it a wild
goose chase to read into it too much)

If anyone else has any ideas I would be much appreciated. Elman Thankyou
very much for taking the time out to respond.

Would any of you suggest blatting the PC and starting again with a basic
bare minimum config, i.e. no GUI 

Regards
 
Stuart Buchanan
 
 
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-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Elman Efendiyev
Sent: 24 July 2004 19:53
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] Please help I fear I have missed something
very important! but what?

Looks like you missed 's' extension for incoming calls
You need something like this in extensions.conf

exten => s,1,Answer
exten => s,2,Dial(SIP/1001,20,t)
 
See sample of extensions.conf in asterisk distribution (make samples if
you didn't install samples)

--
Sincerely,
Elman Efendiyev
elman at earlinvest.com 

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Sales
Sent: Saturday, July 24, 2004 9:03 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Please help I fear I have missed something
very important! but what?


Sorry about this, I have been struggling with the basics of my asterisk
config.
 
I set up two sip peers and two phones. And I set up lots of dial masks
for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming
calls to work. So I have gone back to a very basic FWD config, with one
phone which as far as I am aware should work, but doesn't. I cannot find
info on how to fix this.
 
Below is my sip.conf
 
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register => 2xxxx:xxxx at fwd.pulver.com/1001
 
[fwd]
type=friend
secret=xxxxxx
username=xxxxxx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xxxxxx
host=dynamic
secret=xxxxxxx
callerid=Home <1001>
dtmfmode=RFC2833
mailbox=1001
context=sip
 
 
and here is my extensions.conf:
 
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten => 1001,2,Dial(SIP/1001,20,t)
include => fwdnet
;
[fwdnet]
exten => _8.,1,Dial,SIP/${EXTEN:1}@fwd,t
 
 
Now as I said I can call out no probs by dialing 8 then the FWD number,
but incoming calls don't work, and as far as I can see that should ring
ext 1001 for 20 secs.
 
Could someone please help a complete Linux/Asterisk Newb, as apart from
this I have learnt a hell of a lot. But it's the last thing I need to
solve.
 
The linux box for this testing has a unfirewalled public IP address, so
there is no problems with NAT
 
Please please can someone help. If I have missed something important
then I aplogise, as I have been scouring the wiki and the archives to no
avail
 
Regards
 
 
Stuart Buchanan
 
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