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

Thu May 22 01:35:36 CDT 2008

Hi Roberto,

i added this exntesions.conf 

[spa]
Exten => _1XXX,1,Dial(SIP/${EXTEN})
exten => _0.,1,Dial(SIP/1009/${EXTEN:1})

and in sip.conf:

[1009]
username=1009
type=friend
secret=1234
host=dynamic
canreinvite=yes
context=spa
disallow=all
allow=alaw
dtmfmode=info
qualify=yes
callgroup=1
pickupgroup=1

which i have the extension 1009 in sip.conf directed to it..
and then tried calling out but it still gave me the same error! 

    -- Executing [01442302 at spa:1] Dial("SIP/1003-b5f0b840", "SIP/1009/01444444") in new stack
    -- Called 1009/01444444
    -- Got SIP response 503 "Service Unavailable" back from 192.168.0.111
    -- SIP/1009-0821d888 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/1003-b5f0b840' status is 'CONGESTION'

From: roberto.milani at sbcglobal.net
To: asterisk-users at lists.digium.com
Date: Wed, 21 May 2008 10:29:21 -0700
Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to	pstn calls)

Ciao Roland
your dialplan:Exten => _1XX,1,Dial(SIP/${EXTEN}) 

_1XX is a three (3) digit number starting with 1, I'm not sure what happens if you dial 1009 but it seems that it is dialing.
Anyway the ${EXTEN} is 1009 so asterisk is trying to dial that extension which doesn't exist.
your dial out should look something like:
[outgoing]
exten => _9.,1,Dial(SIP/100/${EXTEN:1})
where you're specifying that all the calls that starts with 9 will go to extension 100 (assuming that is your spa-3102) and there you dial the number dialed from the caller stripped by the 9 (that is the :1 after EXTEN)Now 9 is standard in USA for outside line, in some other countries is 0, you choose
CiaoRoberto
On May 21, 2008, at 7:52 AM, RoLaNd RoLaNd wrote:

Hello Roberto,

first of all, id like to thank you for your help with this..
secondly, i tried the configuration you gave me but it still gave me the same error..! 
but just to b sure ill tell u wht im doing..
after following ur advice to the letter.. i kept my asterisk configuration the same the only thing i edited in sip.conf is adding the port for the pstn extension to match the one in sipura 3102.. and gave the PSTN line interface on sipura the user id of " 1009"
then i called from my softphone 1009 so i could dial out.. 
and it gave me this error in asterisk cli:

 Connect attempt from '127.0.0.1' unable to authenticate
    -- Executing [1009 at spa:1] Dial("SIP/1003-b5f0e828", "SIP/1009") in new stack
    -- Called 1009
    -- Got SIP response 503 "Service Unavailable" back from 192.168.0.111
    -- SIP/1009-0821d888 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/1003-b5f0e828' status is 'CONGESTION'
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found

is that the right way of doing this?! do i call 1009 (pstn line user id) or wht! 
ps: could us hare with me ur sip.conf and extensions.conf please just to compare mine with urs maybe something is missing! 

once again thanks for ur help :)

> Message: 22
> Date: Wed, 21 May 2008 06:49:39 -0700
> From: Roberto Milani <roberto.milani at sbcglobal.net>
> Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to
> sip/sip to pstn calls)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <D01A8127-5C23-4329-8A5A-4079203B0B99 at sbcglobal.net>
> Content-Type: text/plain; charset="windows-1252"
> 
> 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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