I appreciate all the help and I am learning quite a bit here...
I have started over completely from scratch....
Completely removed ALL asterisk files and installed most recent CVS HEAD
Compiled all and I still can not make an outgoing SIP call via a sip
provider....
Once again.... toward the end of this message I will include my sip.conf
and extensions.conf section detailing the simple dialplan section.....
This is the error I get if I try to dial out:
------------------------------------------------------------------------
Asterisk Ready.
*CLI> -- Executing Dial("SIP/77-d1a3",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
Jun 6 21:41:15 WARNING[8440]: chan_sip.c:8475 handle_response: Forbidden
- wrong password on authentication for INVITE to '"Steve 5.8Ghz Cordless"
<sip:[EMAIL PROTECTED]>;tag=as26492b95'
-- SIP/stanaphone-out-e834 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Got SIP response 481 "Call Leg Does Not Exist" back from
204.147.183.18
-------------------------------------------------------------------------
Still saying something about wrong password.... which makes no sense cause
I am using the correct password.
And just as before and for the last 3 weeks..... register line works fine
and incoming calls work just fine.....
I continue to be unable to make an outbound call with asterisk...
I can plug in another device with same account (IP phone or softphone) and
outbound calls work just fine...
Just been a LONG nogo getting asterisk to do it.
Any more pointers would be greatly appreciated..
Thanks!!!
Steve
------------- sip.conf ---------------
[general]
port = 5060
bindaddr = 0.0.0.0
allow=ulaw
; This section is because i'm behind nat
externip = 68.42.113.92 ;Outside address
localnet = 10.73.73.133 ;Inside address
localmask = 255.255.255.0 ;Inside subnet
context = sip ; Default context for incoming calls
register => 7345551212:[EMAIL PROTECTED]/77
[stanaphone-out]
type = friend
username = 7345551212
;authuser = 7345551212
secret = secretpassword
host = sip.stanaphone.com
nat = yes
canreinvite=no
insecure=very
------------------------------
--------- extensions.conf --------
[sip]
include => default
exten =>_1XXXXXXXXXX,1,Dial(SIP/[EMAIL PROTECTED])
exten => 78,1,Dial(SIP/78,20)
exten => 77,1,Dial(SIP/77,20)
---------------------------------------
Version info:
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-06-06 22:32:05
*CLI> show version files
File Revision
---- --------
cdr_custom.c Revision: 1.11
cdr_manager.c Revision: 1.6
cdr_csv.c Revision: 1.16
pbx_functions.c Revision: 1.3
chan_zap.c Revision: 1.458
chan_phone.c Revision: 1.52
chan_modem_i4l.c Revision: 1.27
chan_oss.c Revision: 1.49
chan_features.c Revision: 1.12
chan_skinny.c Revision: 1.78
chan_local.c Revision: 1.47
chan_iax2.c Revision: 1.303
iax2-parser.c Revision: 1.45
iax2-provision.c Revision: 1.12
chan_mgcp.c Revision: 1.123
chan_agent.c Revision: 1.136
chan_modem_bestdata.c Revision: 1.16
chan_sip.c Revision: 1.754
chan_modem_aopen.c Revision: 1.15
chan_modem.c Revision: 1.40
io.c Revision: 1.10
sched.c Revision: 1.19
logger.c Revision: 1.74
frame.c Revision: 1.57
loader.c Revision: 1.45
config.c Revision: 1.66
channel.c Revision: 1.202
translate.c Revision: 1.37
file.c Revision: 1.68
say.c Revision: 1.60
pbx.c Revision: 1.254
cli.c Revision: 1.86
md5.c Revision: 1.14
term.c Revision: 1.10
ulaw.c Revision: 1.4
alaw.c Revision: 1.3
callerid.c Revision: 1.32
fskmodem.c Revision: 1.7
image.c Revision: 1.15
app.c Revision: 1.66
cdr.c Revision: 1.40
tdd.c Revision: 1.6
acl.c Revision: 1.45
rtp.c Revision: 1.133
manager.c Revision: 1.99
asterisk.c Revision: 1.162
dsp.c Revision: 1.43
chanvars.c Revision: 1.8
indications.c Revision: 1.25
autoservice.c Revision: 1.12
db.c Revision: 1.18
privacy.c Revision: 1.5
enum.c Revision: 1.26
srv.c Revision: 1.13
dns.c Revision: 1.14
utils.c Revision: 1.47
config_old.c Revision: 1.4
plc.c Revision: 1.5
jitterbuf.c Revision: 1.15
dnsmgr.c Revision: 1.5
*CLI>
On Mon, 6 Jun 2005, Steve wrote:
OK,
Thanks!
Didn't realize that :-)
Still learning.
Going to go cleanup and do all over again!
Take care,
Steve
On Mon, 6 Jun 2005, Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Steve <[EMAIL PROTECTED]> wrote:
Still just simply want to be able to make an outbound sip provider call
from asterisk.... that's all :-)
Kinda like that guy that wants to call his girlfriend....
I'm getting lonely here.
Ok completely started over....
Installed CVS-HEAD
zaptel seems to compile ok (see lots of warnings)
libpri seems to compile ok
zaptel and ztdummy load ok after compile
asterisk builds ok but exits with this error at runtime:
[pbx_wilcalu.so]Jun 6 14:32:54 WARNING[27986]: loader.c
:310 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined
symbol: ast_p
thread_create
Jun 6 14:33:49 WARNING[27986]: loader.c:518 load_modules
: Loading module pbx_wilcalu.so failed!
asterisk will not run!
I have no idea what this means or how to deal with it. any help is much
appreciated!
Asterisk version: Vontage:/etc/asterisk# asterisk -V
Asterisk CVS-HEAD
umm.... not really informative there.... :-) I downloaded and built it
June, 6
2:45PM Eastern STD time (US)
Here's some more info about my system just in case it is userful:
Stable compiles & runs OK.
You must make sure you empty out /usr/lib/asterisk/modules any time you
change between installing Head and installing Stable. Some of the .so
file names have changed, and if there are old ones left over it will
confuse things.
Cheers
Tony
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