James FitzGibbon wrote:
> [EMAIL PROTECTED] wrote:
>
>>
>> Well I have a Sipura 3000 connected at my home in Georgetown
>> which takes inbound PSTN calls back to an * server in
>> Brampton (to an IVR menu), the FXS port on this device is an
>> extension of the * server
>>
>> * is able to use the FXO port for outbound PSTN calls in
>> Georgetown if the correct prefix is entered
>>
>> I'll admit it took some playing around, but once configured
>> has worked flawlessly
>>
>>
> What tact did you take in configuring it?
The SPA3000 is a fussy beast, but it turns out there isn/t much
required.
Set it back to factory first.
Then, try these settings in the Sipura:
######## FXS #########
# *** SIP Settings
SIP_Port[1] "5060" ;
# *** Proxy and Registration
Proxy[1] "my.proxy.org" ; Asterisk PBX (eg
10.1.1.2)
# *** Subscriber Information
Display_Name[1] "username1" ; Asterisk setting
User_ID[1] "username1" ; Asterisk setting
Password[1] "password1" ; Asterisk setting
# *** Dial Plan
Dial_Plan[1]
;"([2-9]xx[2-9]xxx|[2-9]xx[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx|011[2-9].|7xx|
7xxx|xx.)" ;
######## FXO #########
# ***
Line_Enable[2] "Yes" ;
# *** Proxy and Registration
Proxy[2] "my.proxy.org" ; Asterisk PBX (eg
10.1.1.2)
# *** Subscriber Information
Display_Name[2] "username2" ; Asterisk setting
User_ID[2] "username2" ; Asterisk setting
Password[2] "password2" ; Asterisk setting
# *** Dial Plans
Dial_Plan_8[2] "(S0<:[EMAIL PROTECTED]>)" ; basically
whatever extension you want calls
; from this
device to start at. I guess wildcards
; would work too
# *** VoIP-To-PSTN Gateway Setup
VoIP-To-PSTN_Gateway_Enable[2] "Yes" ;
VoIP_Caller_Auth_Method[2] "none" ; # options: none/PIN/HTTP
Digest
One_Stage_Dialing[2] "Yes" ;
# *** PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP_Gateway_Enable[2] "Yes" ;
PSTN_Caller_Auth_Method[2] "none" ; # options: none/PIN
PSTN_Ring_Thru_Line_1[2] "No" ; "Yes" will always ring the FXS
PSTN_CID_For_VoIP_CID[2] "Yes" ; CID will be passed from the CO
PSTN_Caller_Default_DP[2] "8" ; # options: 1/2/3/4/5/6/7/8
######## GLOBAL #########
# *** Control Timer Values (sec)
Interdigit_Long_Timer "10" ; perhaps set this to 3
Interdigit_Short_Timer "3" ;
> And were you
> adding it to a
> working * setup, or were you working from scratch? I have
> been trying
> to get a "minimal" asterisk configuration working, but even
> that seems
> to be rather bloated.
modules.conf is the key to a minimal asterisk setup
> Does anyone know what the true minimum configuration for *
> with a single
> SIP friend and voicemail? I would have thought I could get
> away with asterisk.conf, sip.conf, extensions.conf and
> voicemail.conf,
Actually, you can get away with even less than that, if you configure
modules.conf correctly.
> but when I
> try that I get errors from chan_modem.so.
noload=> chan_modem.so
or
; load=> chan_modem.so
> If I add an empty
> version of
> the config for that module, another one comes up an whines.
Yep, so keep remarking them out until you get no more complaints. Takes
five minutes, tops.
> The process
> repeats until by the time * will start up, I no longer have
> anything that could be called minimal.
But if you do it the other way around, you'll achive a very lean system.
> I've tried turning off module autoload and just pulling in
> the modules
> that I think I need, but there seem to be some complex
> inter-dependencies. For example, a few symbols in chan_modem
> are used
> by other modules, even if you don't have any ISDN modem cards
> installed.
Nevertheless, this is not a bad apprach at all. Leif posted a sample
modules.conf on the wiki that sets autoload=no, and then has a load=>
for wach module. I always use autoload=no. I want to control determine
what I load, since I can't see any value in cluttering up my system with
resources I'm not using.
You then just selectively work though each complaint (by remarking out
the offending module) until you have a lean and mean Asterisk running.
Jim.
--
Jim Van Meggelen
[EMAIL PROTECTED]
http://www.oreillynet.com/pub/au/2177
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