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

Wed May 21 12:29:21 CDT 2008

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

Ciao
Roberto

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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