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Carrier is bandwidth.com<br><br>we are running Asterisk 1.6.1.1<br><br>i ran sip set debug on from the CLI<br><br>Once i did a module reload it started displaying all the debuging info. Here is some of the debug info<br><br><br>--- (13 headers 0 lines) ---<br>Scheduling destruction of SIP dialog '5fafc3b57e3928877141d12f58c9f7a2@127.0.0.2' in 32000 ms (Method: REGISTER)<br>[Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s)<br><br><--- SIP read from UDP://127.0.0.1:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060<br>From: <sip:500@dynamic>;tag=as51c22cdd<br>To: <sip:500@dynamic>;tag=as51c22cdd<br>Call-ID: 7e1e2c4c702c5b1619fef3961219273a@127.0.0.2<br>CSeq: 117 REGISTER<br>Server: Asterisk PBX 1.6.1.1<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Expires: 120<br>Contact: <sip:500@127.0.0.1>;expires=120<br>Date: Fri, 10 Jul 2009 10:53:39 GMT<br>Content-Length: 0<br><br><hr id="stopSpelling">Date: Fri, 10 Jul 2009 09:42:31 -0400<br>From: stotaro@asteriskhelpdesk.com<br>To: asterisk-users@lists.digium.com<br>Subject: Re: [asterisk-users] setting up phones<br><br>Who is the carrier? What flavor of Asterisk are you using?<br><br>Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan.<br><br>If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all?<br>
<br>also, change registersip to yes.<br><br>Thanks,<br>Steve<br><br><div class="EC_gmail_quote">On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a>></span> wrote:<br>
<blockquote class="EC_gmail_quote" style="padding-left: 1ex;">
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Here is my physical network.<br><br>We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. <br><br>the other nic in the Asterisk box is plugged into your lan switch<br>
<br>the phones are plugged into the lan switch<br><br><br>I can ping the phones from the Asterisk server. <br><br><hr>Date: Thu, 9 Jul 2009 17:42:43 -0400<br>From: <a href="mailto:stotaro@asteriskhelpdesk.com">stotaro@asteriskhelpdesk.com</a><div class="EC_im">
<br>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br></div><div class="EC_im">Subject: Re: [asterisk-users] setting up phones<br><br><br><br></div><div><div></div><div class="h5">
<div>On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose <span dir="ltr"><<a href="mailto:sixfourimpala@hotmail.com">sixfourimpala@hotmail.com</a>></span> wrote:<br><blockquote style="padding-left: 1ex;">
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I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf<br>
</div></blockquote><div><br>Either you did not explain your network topology very well or that is your problem.<br><br>Unless you are trying to segregate your VoIP traffic, plug everything into the switch.<br><br>If using DHCP, get the IP and try pinging the phones from the Asterisk box.<br>
<br>I bet it is just a network issue. <br></div></div><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br><br></div></div><div class="EC_im"><hr>Windows Live™: Keep your life in sync. <a href="http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009">Check it out.</a></div>
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