[asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)

Wed May 21 08:49:39 CDT 2008

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:

> Hello all,
>
> its been a while im trying to setup my asterisk/sipura 3102 to  
> recieve/make calls from softphones on pcs in my home..
> i've set up 5 SIP extensions in sip.conf and made the dialing plan  
> in extensions.conf..
> i could make calls from 1 sip phone to another in my home.. but i  
> cant call out using pstn line interface nor recieve calls..
> please find below my topology as well as config info:
>
>                          (192.168.0.0)
>        ____________LAN______________
>       |                        |                   |
> softphone              asterisk           sipura---------PSTN LINE
>
>
>
> Configuration:
>
> ASTERISK:
>
> sip.conf
>
> [101]
> type=peer
> port=5062
> host=dynamic
> secret=1234
> context=spa
>
>
> [103]
> type=peer
> port=5061
> host=dynamic
> secret=1234
> context=spa
>
> [100]
> type=peer
> port=5061
> host=dynamic
> secret=1234
> context=spa
>
> [111]
> type=peer
> port=5060
> host=dynamic
> secret=1234
> context=spa
>
> ================================================== ===========
>
> EXTENSIONS.CONF
>
> [spa]
> Exten => _1XX,1,Dial(SIP/${EXTEN})
>
> ================================================== ===========
>
>
> and this is the settings i have right now for sipura 3102 in my PSTN  
> LINE:
>
>
> http://img84.imageshack.us/my.php?image=40541922um2.jpg
>
> http://img98.imageshack.us/my.php?image=55448347ss9.jpg
>
> http://img262.imageshack.us/my.php?imag ... 472qz3.jpg
>
> ps: i read so many tutorials and none seems to help..
> lately whenever i try to call out using my sipphone.. it gives me  
> "503 service unavailable" and this is wht shows on the CLI of  
> asterisk when i set sip debug on..
>
>
>
>
> ubuntu-pbx-desktop*CLI>
>   == Connect attempt from '127.0.0.1' unable to authenticate
>     -- Executing [1009 at spa:1] Dial("SIP/1003-b5f05600", "SIP/1009")  
> in new stack
>     -- Called 1009*CLI>
>     -- Got SIP response 410 "Gone" back from 192.168.0.111
>     -- SIP/1009-081741d0 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
>   == Auto fallthrough, channel 'SIP/1003-b5f05600' status is  
> 'CONGESTION'
>
>
>
> Invite your mail contacts to join your friends list with Windows  
> Live Spaces. It's easy! Try it! 
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