[Asterisk-Users] Some SIP Setup problems

[Mike Nash]

Hi

I'm trying to configure my Asterisk box to provide a simple sample 
configuration.  It's a mandrake 9.1 box, no cards except a sound card.  The 
config I am trying to achieve is simply one server, with two SIP clients.

Two issues are cropping up - the first, when I start Asterisk, the sound goes 
nuts and I get an error (below)

Jan 25 18:16:44 WARNING[163851]: chan_oss.c:268 sound_thread: Read error on 
sound device: Resource temporarily unavailable

When Asterisk starts, I get the error (below):

Jan 25 18:06:33 WARNING[81926]: chan_sip.c:446 __sip_xmit: sip_xmit of 
0x80db77c (len 459) to 0.0.0.0 returned -1: Invalid argument

I'm pretty confident this second error is because I have misconfigured 
extensions.conf and sip.conf, but I can't see why.   When I try to connect to 
the server with an XTEN client, I get this error:

Jan 25 18:15:38 NOTICE[81926]: chan_sip.c:5548 handle_request: Registration 
from 'Mike <sip:mike at 203.109.242.119>' failed for '203.118.186.16'

I've tried looking at the www.automated.it setup information, along with the 
information on fnords.org - this has gotten me this far, but I can't see for 
the life of me what I have done wrong.

If anyone could provide me some pointers, it would be much appreciated.

Regards


Mike

My SIP conf looks like this:

;
; SIP Configuration for Asterisk
;
[general]
port = 5060			; Port to bind to
bindaddr = 0.0.0.0		; Address to bind to
context = sip	; Default for incoming calls

[Phone1]
type=friend
secret=yap
auth=md5
nat=yes
host=dynamic
dtfmmode=inband
mailbox=1000
username=mike
context=sip
disallow=all
allow=gsm
callerid="Mike Nash" <6969>

[Phone2]
type=friend
secret=yap
auth=md5
nat=yes
host=dynamic
dtfmmode=inband
mailbox=1000
username=darryl
context=sip
disallow=all
allow=gsm
callerid="Darryl West" <6970>



My extensions.conf looks like this:

[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; For more information on applications, just type "show applications" at your
; friendly Asterisk CLI prompt.
;
[sip]
exten => 1,1,Dial(SIP/Phone1,20,tr)
exten => 2,1,Dial(SIP/Phone2,20,tr)
exten => 1000,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)