VoIP Mailing List Archives

[roberto.milani at sbcg...]

Hi Roland

I have 2 linksys spa-3102 working pretty good both dialing in and out
and I followed this instructions to set it up:
update to the latest firmware then:

..Go to the first tab ?Voice? and sixth sub-tab ?Line 1?
....SIP Settings:
......SIP Port: Notice that it is set to 5060 for line 1 and 5061 for
PSTN Line (next tab). These port values must be correctly transferred
to the correct contexts in sip.conf.
....Proxy and registration:
......Proxy: 192.168.5.70 < The IP address of your Asterisk server
....Subscriber Information:
......Display Name: LivingRoom < This will be the test phone, but any
name would do as lone as it is used in the configuration files.
......User ID: LivingRoom
......Password: SomePassword
......Auth ID: LivingRoom < probably not needed
....Dial Plan:
......Dial Plan: (*xx|[3469]11|0|00|[2-9]xxxxxxxxx|
1xxx[2-9]xxxxxxxxxS0|xxxxxxxxxxxx.) < We have 10 digit local dialing.
The default is set for seven digit local dialing. Adjust as needed.
......Emergency Number: < Hmmm, I don?t know what to do here: it?s
probably important, but it is poor form to dial 911 just to test. . .
Help?
....Click Submit All Changes

..Go to the first tab ?Voice? and seventh sub-tab ?PSTN?:
....SIP Settings:
......SIP Port: Notice that it is set to 5061 for PSTN User and 5060
for Line 1. These port values must be correctly transferred to the
correct contexts in sip.conf.
....Proxy and Registration:
......Proxy: 192.168.5.70 < The IP address of your Asterisk server
....Subscriber Information:
......Display Name: PSTN1 < I have two lines so there is an PSTN2, but
we will not discuss it here.
......User ID: PSTN1
......Password: SomePassword
......Auth ID: PSTN1 < probably not needed.
....Dial Plans:
......Dial Plan 2: (S0<:PSTN1>) < That is an S-zero. The incoming call
will be passed to your extensions.conf file with extension ?PSTN1?
where we will Playback a greeting to the caller and then playback the
main menu of our internal users and their extension numbers. You can
also use specific extension numbers, such as: (S0<:2091>), which will
send all incoming calls to that extension for processing. This might
work best with two or more external lines where a second call comes in
while the first is being processed through the main menu and extension
capture.
....VoIP-To-PSTN Gateway Setup:
......Line 1 VoIP Caller DP: 1 < Leave this at 1. The SPA3102 will use
the Dial Plan 1 (above = (xx.)) so all your Dial Plan decision making
will be done in the Asterisk extensions.conf file. The SPA3102 will
dial out whatever Asterisk hands out.
....PSTN-To-VoIP Gateway Setup:
......PSTN Ring Thru Line 1: no < When this is ?yes?, an incoming call
goes directly through to Line 1. We only want line 1 to ring when
Asterisk routs a call to it.
......PSTN CID for VoIP CID: yes < capture the Caller ID provided by
the incoming call and pass it through to Asterisk to display on your
internal phones.
......PSTN Caller Default DP: 2 < Change to 2. The incoming call will
be passed to your extensions.conf file with extension 's' as defined
in Dial Plan 2 (above).
......Off Hook While Calling VoIP: no < I read this in some Google
search. I don?t know what it does, but stuff seems to work. Help?
....FXO Timer Values (sec):
......PSTN Answer Delay: 5 < Delay so that you can get the CID data.
NghtShd at http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html
claims that 5 seconds is long enough.
....Click Submit All Changes

Ciao

Roberto

On May 21, 2008, at 6:00 AM, RoLaNd RoLaNd wrote: