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

[RoLaNd RoLaNd]

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! > > _______________________________________________> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --> >> > asterisk-users mailing list> > To UNSUBSCRIBE or update options visit:> > http://lists.digium.com/mailman/listinfo/asterisk-users> > -------------- next part --------------> An HTML attachment was scrubbed...> URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080521/7c9ef721/attachment.htm > > ------------------------------> > _______________________________________________> --Bandwidth and Colocation Provided by http://www.api-digital.com--> > asterisk-users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users> > End of asterisk-users Digest, Vol 46, Issue 69> **********************************************
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