<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi Roland<div><br></div><div>I have 2 linksys spa-3102 working pretty good both dialing in and out and I followed this instructions to set it up:</div><div><br></div><div><br></div><div>update to the latest firmware then:</div><div><br></div><div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">..Go to the first tab ‘Voice’ and sixth sub-tab ‘Line 1’ </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....SIP Settings: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Proxy and registration: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Proxy: 192.168.5.70 < The IP address of your Asterisk server </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Subscriber Information: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Display Name: LivingRoom < This will be the test phone, but any name would do as lone as it is used in the configuration files. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......User ID: LivingRoom </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Password: SomePassword </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Auth ID: LivingRoom < probably not needed </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Dial Plan: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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? </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Click Submit All Changes </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; min-height: 12px; "><span style="letter-spacing: 0.0px"></span><br></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">..Go to the first tab ‘Voice’ and seventh sub-tab ‘PSTN’: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....SIP Settings: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Proxy and Registration: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Proxy: 192.168.5.70 < The IP address of your Asterisk server </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Subscriber Information: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Display Name: PSTN1 < I have two lines so there is an PSTN2, but we will not discuss it here. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......User ID: PSTN1 </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Password: SomePassword </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......Auth ID: PSTN1 < probably not needed. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Dial Plans: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....VoIP-To-PSTN Gateway Setup: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....PSTN-To-VoIP Gateway Setup: </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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). </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......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? </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....FXO Timer Values (sec): </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">......PSTN Answer Delay: 5 < Delay so that you can get the CID data. NghtShd at <a href="http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html"><span style="text-decoration: underline ; letter-spacing: 0.0px color: #000099">http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html</span></a> claims that 5 seconds is long enough. </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><span style="letter-spacing: 0.0px">....Click Submit All Changes </span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><br></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; ">Ciao</div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><br></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; ">Roberto</div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; line-height: 18px; font: normal normal normal 10px/normal Helvetica; "><br></div><div><div>On May 21, 2008, at 6:00 AM, RoLaNd RoLaNd wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="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; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0; "><div class="hmmessage" style="font-size: 10pt; font-family: Tahoma; ">Hello all,<br> <br>its been a while im trying to setup my asterisk/sipura 3102 to 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 in extensions.conf..<br>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..<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:<span class="Apple-converted-space"> </span><br><br>sip.conf<span class="Apple-converted-space"> </span><br><br>[101]<span class="Apple-converted-space"> </span><br>type=peer<span class="Apple-converted-space"> </span><br>port=5062<span class="Apple-converted-space"> </span><br>host=dynamic<span class="Apple-converted-space"> </span><br>secret=1234<span class="Apple-converted-space"> </span><br>context=spa<span class="Apple-converted-space"> </span><br><br><br>[103]<span class="Apple-converted-space"> </span><br>type=peer<span class="Apple-converted-space"> </span><br>port=5061<span class="Apple-converted-space"> </span><br>host=dynamic<span class="Apple-converted-space"> </span><br>secret=1234<span class="Apple-converted-space"> </span><br>context=spa<span class="Apple-converted-space"> </span><br><br>[100]<span class="Apple-converted-space"> </span><br>type=peer<span class="Apple-converted-space"> </span><br>port=5061<span class="Apple-converted-space"> </span><br>host=dynamic<span class="Apple-converted-space"> </span><br>secret=1234<span class="Apple-converted-space"> </span><br>context=spa<span class="Apple-converted-space"> </span><br><br>[111]<span class="Apple-converted-space"> </span><br>type=peer<span class="Apple-converted-space"> </span><br>port=5060<span class="Apple-converted-space"> </span><br>host=dynamic<span class="Apple-converted-space"> </span><br>secret=1234<span class="Apple-converted-space"> </span><br>context=spa<span class="Apple-converted-space"> </span><br><br>================================================== ===========<span class="Apple-converted-space"> </span><br><br>EXTENSIONS.CONF<span class="Apple-converted-space"> </span><br><br>[spa]<span class="Apple-converted-space"> </span><br>Exten => _1XX,1,Dial(SIP/${EXTEN})<span class="Apple-converted-space"> </span><br><br>================================================== ===========<span class="Apple-converted-space"> </span><br><br><br>and this is the settings i have right now for sipura 3102 in my PSTN LINE:<span class="Apple-converted-space"> </span><br><br><br><a onclick="urchinTracker ('/outgoing/http_www_voipuser_org_ship_to_php_url_http_img84_imageshack_us_my_php_image_40541922um2_jpg');" href="http://www.voipuser.org/ship_to.php?url=http://img84.imageshack.us/my.php?image=40541922um2.jpg" target="_blank"><u><font face="" color="#0000ff">http://img84.imageshack.us/my.php?image=40541922um2.jpg</font></u></a><span class="Apple-converted-space"> </span><br><br><a onclick="urchinTracker ('/outgoing/http_www_voipuser_org_ship_to_php_url_http_img98_imageshack_us_my_php_image_55448347ss9_jpg');" href="http://www.voipuser.org/ship_to.php?url=http://img98.imageshack.us/my.php?image=55448347ss9.jpg" target="_blank"><u><font face="" color="#0000ff">http://img98.imageshack.us/my.php?image=55448347ss9.jpg</font></u></a><span class="Apple-converted-space"> </span><br><br><a onclick="urchinTracker ('/outgoing/http_www_voipuser_org_ship_to_php_url_http_img262_imageshack_us_my_php_image_43929472qz3_jpg');" href="http://img262.imageshack.us/my.php?imag ... 472qz3.jpg" target="_blank"><u><font face="" color="#0000ff">http://img262.imageshack.us/my.php?imag ... 472qz3.jpg</font></u></a><span class="Apple-converted-space"> </span><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 "503 service unavailable" and this is wht shows on the CLI of 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") 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 'CONGESTION'<br><br> <br><br><hr>Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!<span class="Apple-converted-space"> </span><a href="http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us" target="_new">Try it!</a>_______________________________________________<br>-- Bandwidth and Colocation Provided by<span class="Apple-converted-space"> </span><a href="http://www.api-digital.com">http://www.api-digital.com</a><span class="Apple-converted-space"> </span>--<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></span></blockquote></div><br></div></body></html>