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<body class='hmmessage'><BR><BR>Hello Roberto,<BR>
<BR>
first of all, id like to thank you for your help with this..<BR>
secondly, i tried the configuration you gave me but it still gave me the same error..! <BR>
but just to b sure ill tell u wht im doing..<BR>
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"<BR>
then i called from my softphone 1009 so i could dial out.. <BR>
and it gave me this error in asterisk cli:<BR>
<BR>
<BR>
Connect attempt from '127.0.0.1' unable to authenticate<BR> -- Executing [1009@spa:1] Dial("SIP/1003-b5f0e828", "SIP/1009") in new stack<BR> -- Called 1009<BR> -- Got SIP response 503 "Service Unavailable" back from 192.168.0.111<BR> -- SIP/1009-0821d888 is circuit-busy<BR> == Everyone is busy/congested at this time (1:0/1/0)<BR> == Auto fallthrough, channel 'SIP/1003-b5f0e828' status is 'CONGESTION'<BR> == Parsing '/etc/asterisk/manager.conf': Found<BR> == Parsing '/etc/asterisk/manager.d/op-panel.conf': Found<BR> == Parsing '/etc/asterisk/users.conf': Found<BR>
<BR>
<BR>is that the right way of doing this?! do i call 1009 (pstn line user id) or wht! <BR>
ps: could us hare with me ur sip.conf and extensions.conf please just to compare mine with urs maybe something is missing! <BR>
<BR>
once again thanks for ur help :)<BR><BR>
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<BR>> Message: 22<BR>> Date: Wed, 21 May 2008 06:49:39 -0700<BR>> From: Roberto Milani <roberto.milani@sbcglobal.net><BR>> Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to<BR>> sip/sip to pstn calls)<BR>> To: Asterisk Users Mailing List - Non-Commercial Discussion<BR>> <asterisk-users@lists.digium.com><BR>> Message-ID: <D01A8127-5C23-4329-8A5A-4079203B0B99@sbcglobal.net><BR>> Content-Type: text/plain; charset="windows-1252"<BR>> <BR>> Hi Roland<BR>> <BR>> I have 2 linksys spa-3102 working pretty good both dialing in and out <BR>> and I followed this instructions to set it up:<BR>> <BR>> <BR>> update to the latest firmware then:<BR>> <BR>> ..Go to the first tab ?Voice? and sixth sub-tab ?Line 1?<BR>> ....SIP Settings:<BR>> ......SIP Port: Notice that it is set to 5060 for line 1 and 5061 for <BR>> PSTN Line (next tab). These port values must be correctly transferred <BR>> to the correct contexts in sip.conf.<BR>> ....Proxy and registration:<BR>> ......Proxy: 192.168.5.70 < The IP address of your Asterisk server<BR>> ....Subscriber Information:<BR>> ......Display Name: LivingRoom < This will be the test phone, but any <BR>> name would do as lone as it is used in the configuration files.<BR>> ......User ID: LivingRoom<BR>> ......Password: SomePassword<BR>> ......Auth ID: LivingRoom < probably not needed<BR>> ....Dial Plan:<BR>> ......Dial Plan: (*xx|[3469]11|0|00|[2-9]xxxxxxxxx| <BR>> 1xxx[2-9]xxxxxxxxxS0|xxxxxxxxxxxx.) < We have 10 digit local dialing. <BR>> The default is set for seven digit local dialing. Adjust as needed.<BR>> ......Emergency Number: < Hmmm, I don?t know what to do here: it?s <BR>> probably important, but it is poor form to dial 911 just to test. . . <BR>> Help?<BR>> ....Click Submit All Changes<BR>> <BR>> ..Go to the first tab ?Voice? and seventh sub-tab ?PSTN?:<BR>> ....SIP Settings:<BR>> ......SIP Port: Notice that it is set to 5061 for PSTN User and 5060 <BR>> for Line 1. These port values must be correctly transferred to the <BR>> correct contexts in sip.conf.<BR>> ....Proxy and Registration:<BR>> ......Proxy: 192.168.5.70 < The IP address of your Asterisk server<BR>> ....Subscriber Information:<BR>> ......Display Name: PSTN1 < I have two lines so there is an PSTN2, but <BR>> we will not discuss it here.<BR>> ......User ID: PSTN1<BR>> ......Password: SomePassword<BR>> ......Auth ID: PSTN1 < probably not needed.<BR>> ....Dial Plans:<BR>> ......Dial Plan 2: (S0<:PSTN1>) < That is an S-zero. The incoming call <BR>> will be passed to your extensions.conf file with extension ?PSTN1? <BR>> where we will Playback a greeting to the caller and then playback the <BR>> main menu of our internal users and their extension numbers. You can <BR>> also use specific extension numbers, such as: (S0<:2091>), which will <BR>> send all incoming calls to that extension for processing. This might <BR>> work best with two or more external lines where a second call comes in <BR>> while the first is being processed through the main menu and extension <BR>> capture.<BR>> ....VoIP-To-PSTN Gateway Setup:<BR>> ......Line 1 VoIP Caller DP: 1 < Leave this at 1. The SPA3102 will use <BR>> the Dial Plan 1 (above = (xx.)) so all your Dial Plan decision making <BR>> will be done in the Asterisk extensions.conf file. The SPA3102 will <BR>> dial out whatever Asterisk hands out.<BR>> ....PSTN-To-VoIP Gateway Setup:<BR>> ......PSTN Ring Thru Line 1: no < When this is ?yes?, an incoming call <BR>> goes directly through to Line 1. We only want line 1 to ring when <BR>> Asterisk routs a call to it.<BR>> ......PSTN CID for VoIP CID: yes < capture the Caller ID provided by <BR>> the incoming call and pass it through to Asterisk to display on your <BR>> internal phones.<BR>> ......PSTN Caller Default DP: 2 < Change to 2. The incoming call will <BR>> be passed to your extensions.conf file with extension 's' as defined <BR>> in Dial Plan 2 (above).<BR>> ......Off Hook While Calling VoIP: no < I read this in some Google <BR>> search. I don?t know what it does, but stuff seems to work. Help?<BR>> ....FXO Timer Values (sec):<BR>> ......PSTN Answer Delay: 5 < Delay so that you can get the CID data. <BR>> NghtShd at http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html <BR>> claims that 5 seconds is long enough.<BR>> ....Click Submit All Changes<BR>> <BR>> Ciao<BR>> <BR>> Roberto<BR>> <BR>> On May 21, 2008, at 6:00 AM, RoLaNd RoLaNd wrote:<BR>> <BR>> > Hello all,<BR>> ><BR>> > its been a while im trying to setup my asterisk/sipura 3102 to <BR>> > recieve/make calls from softphones on pcs in my home..<BR>> > i've set up 5 SIP extensions in sip.conf and made the dialing plan <BR>> > in extensions.conf..<BR>> > i could make calls from 1 sip phone to another in my home.. but i <BR>> > cant call out using pstn line interface nor recieve calls..<BR>> > please find below my topology as well as config info:<BR>> ><BR>> > (192.168.0.0)<BR>> > ____________LAN______________<BR>> > | | |<BR>> > softphone asterisk sipura---------PSTN LINE<BR>> ><BR>> ><BR>> ><BR>> > Configuration:<BR>> ><BR>> > ASTERISK:<BR>> ><BR>> > sip.conf<BR>> ><BR>> > [101]<BR>> > type=peer<BR>> > port=5062<BR>> > host=dynamic<BR>> > secret=1234<BR>> > context=spa<BR>> ><BR>> ><BR>> > [103]<BR>> > type=peer<BR>> > port=5061<BR>> > host=dynamic<BR>> > secret=1234<BR>> > context=spa<BR>> ><BR>> > [100]<BR>> > type=peer<BR>> > port=5061<BR>> > host=dynamic<BR>> > secret=1234<BR>> > context=spa<BR>> ><BR>> > [111]<BR>> > type=peer<BR>> > port=5060<BR>> > host=dynamic<BR>> > secret=1234<BR>> > context=spa<BR>> ><BR>> > ================================================== ===========<BR>> ><BR>> > EXTENSIONS.CONF<BR>> ><BR>> > [spa]<BR>> > Exten => _1XX,1,Dial(SIP/${EXTEN})<BR>> ><BR>> > ================================================== ===========<BR>> ><BR>> ><BR>> > and this is the settings i have right now for sipura 3102 in my PSTN <BR>> > LINE:<BR>> ><BR>> ><BR>> > http://img84.imageshack.us/my.php?image=40541922um2.jpg<BR>> ><BR>> > http://img98.imageshack.us/my.php?image=55448347ss9.jpg<BR>> ><BR>> > http://img262.imageshack.us/my.php?imag ... 472qz3.jpg<BR>> ><BR>> > ps: i read so many tutorials and none seems to help..<BR>> > lately whenever i try to call out using my sipphone.. it gives me <BR>> > "503 service unavailable" and this is wht shows on the CLI of <BR>> > asterisk when i set sip debug on..<BR>> ><BR>> ><BR>> ><BR>> ><BR>> > ubuntu-pbx-desktop*CLI><BR>> > == Connect attempt from '127.0.0.1' unable to authenticate<BR>> > -- Executing [1009@spa:1] Dial("SIP/1003-b5f05600", "SIP/1009") <BR>> > in new stack<BR>> > -- Called 1009*CLI><BR>> > -- Got SIP response 410 "Gone" back from 192.168.0.111<BR>> > -- SIP/1009-081741d0 is circuit-busy<BR>> > == Everyone is busy/congested at this time (1:0/1/0)<BR>> > == Auto fallthrough, channel 'SIP/1003-b5f05600' status is <BR>> > 'CONGESTION'<BR>> ><BR>> ><BR>> ><BR>> > Invite your mail contacts to join your friends list with Windows <BR>> > Live Spaces. It's easy! Try it! <BR>> > _______________________________________________<BR>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --<BR>> ><BR>> > asterisk-users mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> <BR>> -------------- next part --------------<BR>> An HTML attachment was scrubbed...<BR>> URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080521/7c9ef721/attachment.htm <BR>> <BR>> ------------------------------<BR>> <BR>> _______________________________________________<BR>> --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> <BR>> asterisk-users mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> <BR>> End of asterisk-users Digest, Vol 46, Issue 69<BR>> **********************************************<BR><BR><br /><hr />Get news, entertainment and everything you care about at Live.com. <a href='http://www.live.com/getstarted.aspx ' target='_new'>Check it out!</a></body>
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