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<body class='hmmessage'><div style="text-align: left;">Hi Roberto,<br><br>i added this exntesions.conf <br><br>[spa]<br><span class="EC_Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;">Exten => _1XXX,1,Dial(SIP/${EXTEN})</span><br><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">exten => _0.,1,Dial(SIP/1009/${EXTEN:1})<br><br>and in sip.conf:<br><br>[1009]<br>username=1009<br>type=friend<br>secret=1234<br>host=dynamic<br>canreinvite=yes<br>context=spa<br>disallow=all<br>allow=alaw<br>dtmfmode=info<br>qualify=yes<br>callgroup=1<br>pickupgroup=1<br><br><br><br></span></font>which i have the extension 1009 in sip.conf directed to it..<br>and then tried calling out but it still gave me the same error! <br><br> -- Executing [01442302@spa:1] Dial("SIP/1003-b5f0b840", "SIP/1009/01444444") in new stack<br> -- Called 1009/01444444<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-b5f0b840' status is 'CONGESTION'<br></div><br><br><br><blockquote><hr id="EC_stopSpelling">From: roberto.milani@sbcglobal.net<br>To: asterisk-users@lists.digium.com<br>Date: Wed, 21 May 2008 10:29:21 -0700<br>Subject: Re: [asterisk-users] asterisk and sipura 3102 (pstn to sip/sip to pstn calls)<br><br><div>Ciao Roland</div><div><br></div><div>your dialplan:</div><div><span class="EC_Apple-style-span" style="font-family: Tahoma; font-size: 13px;">Exten => _1XX,1,Dial(SIP/${EXTEN}) <br></span></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;"><br></span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">_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.</span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;"><br></span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">Anyway the ${EXTEN} is 1009 so asterisk is trying to dial that extension which doesn't exist.</span></font></div><div><br></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">your dial out should look something like:</span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;"><br></span></font></div><div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">[outgoing]</span></font></div><div><br></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">exten => _9.,1,Dial(SIP/100/${EXTEN:1})</span></font></div><div><br></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">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)</span></font></div></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">Now 9 is standard in USA for outside line, in some other countries is 0, you choose</span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;"><br></span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">Ciao</span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;">Roberto</span></font></div><div><font class="EC_Apple-style-span" face="Tahoma" size="3"><span class="EC_Apple-style-span" style="font-size: 13px;"><br></span></font></div><div><div>On May 21, 2008, at 7:52 AM, RoLaNd RoLaNd wrote:</div><br class="EC_Apple-interchange-newline"><blockquote><span class="EC_Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div class="EC_hmmessage" style="font-size: 10pt; font-family: Tahoma;"><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..!<span class="EC_Apple-converted-space"> </span><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..<span class="EC_Apple-converted-space"> </span><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!<span class="EC_Apple-converted-space"> </span><br>ps: could us hare with me ur sip.conf and extensions.conf please just to compare mine with urs maybe something is missing!<span class="EC_Apple-converted-space"> </span><br> <br>once again thanks for ur help :)<br><br> <br> <br> <br> <br> <br> <br> <br><br>> Message: 22<br>> Date: Wed, 21 May 2008 06:49:39 -0700<br>> From: Roberto Milani <<a href="mailto:roberto.milani@sbcglobal.net">roberto.milani@sbcglobal.net</a>><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>> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>> Message-ID: <<a href="mailto:D01A8127-5C23-4329-8A5A-4079203B0B99@sbcglobal.net">D01A8127-5C23-4329-8A5A-4079203B0B99@sbcglobal.net</a>><br>> Content-Type: text/plain; charset="windows-1252"<br>><span class="EC_Apple-converted-space"> </span><br>> Hi Roland<br>><span class="EC_Apple-converted-space"> </span><br>> I have 2 linksys spa-3102 working pretty good both dialing in and out<span class="EC_Apple-converted-space"> </span><br>> and I followed this instructions to set it up:<br>><span class="EC_Apple-converted-space"> </span><br>><span class="EC_Apple-converted-space"> </span><br>> update to the latest firmware then:<br>><span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> PSTN Line (next tab). These port values must be correctly transferred<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><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|<span class="EC_Apple-converted-space"> </span><br>> 1xxx[2-9]xxxxxxxxxS0|xxxxxxxxxxxx.) < We have 10 digit local dialing.<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> probably important, but it is poor form to dial 911 just to test. . .<span class="EC_Apple-converted-space"> </span><br>> Help?<br>> ....Click Submit All Changes<br>><span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> for Line 1. These port values must be correctly transferred to the<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> will be passed to your extensions.conf file with extension ?PSTN1?<span class="EC_Apple-converted-space"> </span><br>> where we will Playback a greeting to the caller and then playback the<span class="EC_Apple-converted-space"> </span><br>> main menu of our internal users and their extension numbers. You can<span class="EC_Apple-converted-space"> </span><br>> also use specific extension numbers, such as: (S0<:2091>), which will<span class="EC_Apple-converted-space"> </span><br>> send all incoming calls to that extension for processing. This might<span class="EC_Apple-converted-space"> </span><br>> work best with two or more external lines where a second call comes in<span class="EC_Apple-converted-space"> </span><br>> while the first is being processed through the main menu and extension<span class="EC_Apple-converted-space"> </span><br>> capture.<br>> ....VoIP-To-PSTN Gateway Setup:<br>> ......Line 1 VoIP Caller DP: 1 < Leave this at 1. The SPA3102 will use<span class="EC_Apple-converted-space"> </span><br>> the Dial Plan 1 (above = (xx.)) so all your Dial Plan decision making<span class="EC_Apple-converted-space"> </span><br>> will be done in the Asterisk extensions.conf file. The SPA3102 will<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> goes directly through to Line 1. We only want line 1 to ring when<span class="EC_Apple-converted-space"> </span><br>> Asterisk routs a call to it.<br>> ......PSTN CID for VoIP CID: yes < capture the Caller ID provided by<span class="EC_Apple-converted-space"> </span><br>> the incoming call and pass it through to Asterisk to display on your<span class="EC_Apple-converted-space"> </span><br>> internal phones.<br>> ......PSTN Caller Default DP: 2 < Change to 2. The incoming call will<span class="EC_Apple-converted-space"> </span><br>> be passed to your extensions.conf file with extension 's' as defined<span class="EC_Apple-converted-space"> </span><br>> in Dial Plan 2 (above).<br>> ......Off Hook While Calling VoIP: no < I read this in some Google<span class="EC_Apple-converted-space"> </span><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.<span class="EC_Apple-converted-space"> </span><br>> NghtShd at<span class="EC_Apple-converted-space"> </span><a href="http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html" target="_blank">http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html</a><span class="EC_Apple-converted-space"> </span><br>> claims that 5 seconds is long enough.<br>> ....Click Submit All Changes<br>><span class="EC_Apple-converted-space"> </span><br>> Ciao<br>><span class="EC_Apple-converted-space"> </span><br>> Roberto<br>><span class="EC_Apple-converted-space"> </span><br>> On May 21, 2008, at 6:00 AM, RoLaNd RoLaNd wrote:<br>><span class="EC_Apple-converted-space"> </span><br>> > Hello all,<br>> ><br>> > its been a while im trying to setup my asterisk/sipura 3102 to<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> > in extensions.conf..<br>> > i could make calls from 1 sip phone to another in my home.. but i<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> > LINE:<br>> ><br>> ><br>> ><span class="EC_Apple-converted-space"> </span><a href="http://img84.imageshack.us/my.php?image=40541922um2.jpg" target="_blank">http://img84.imageshack.us/my.php?image=40541922um2.jpg</a><br>> ><br>> ><span class="EC_Apple-converted-space"> </span><a href="http://img98.imageshack.us/my.php?image=55448347ss9.jpg" target="_blank">http://img98.imageshack.us/my.php?image=55448347ss9.jpg</a><br>> ><br>> ><span class="EC_Apple-converted-space"> </span><a href="http://img262.imageshack.us/my.php?imag" target="_blank">http://img262.imageshack.us/my.php?imag</a><span class="EC_Apple-converted-space"> </span>... 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<span class="EC_Apple-converted-space"> </span><br>> > "503 service unavailable" and this is wht shows on the CLI of<span class="EC_Apple-converted-space"> </span><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")<span class="EC_Apple-converted-space"> </span><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<span class="EC_Apple-converted-space"> </span><br>> > 'CONGESTION'<br>> ><br>> ><br>> ><br>> > Invite your mail contacts to join your friends list with Windows<span class="EC_Apple-converted-space"> </span><br>> > Live Spaces. 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