[Asterisk-Users] Re: CLUELESS NEWBIE needs help making an outbound
sip call to PSTN
Steve
asterisk at michiganbroadband.com
Mon Jun 6 18:54:15 MST 2005
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/15168372973 at stanaphone-out") in new stack
-- Called 15168372973 at stanaphone-out
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:77 at 68.42.113.92>;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:secretpassword at sip.stanaphone.com/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/${EXTEN}@stanaphone-out)
exten => 78,1,Dial(SIP/78,20)
exten => 77,1,Dial(SIP/77,20)
---------------------------------------
Version info:
Asterisk CVS-HEAD built by root at Vontage 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 <Pine.LNX.4.61.0506061405510.26398 at mail.michiganbroadband.com>,
>> Steve <asterisk at michiganbroadband.com> 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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