I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text "IAX/SIP registration", but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. <div>
<br></div><div>Thanks for your help.</div><div>naren</div><div><br><br><div class="gmail_quote">On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Did you read the “IAX/SIP registration” section (under Authentication) on <a href="http://voip.ms" target="_blank">voip.ms</a>? <u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>naren<br>
<b>Sent:</b> Tuesday, September 13, 2011 2:22 PM<br><b>To:</b> John Novack<br><b>Cc:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Question about <a href="http://voip.ms" target="_blank">voip.ms</a> service.<u></u><u></u></span></p>
<div><div></div><div class="h5"><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Ok... this is probably a dumb question but I can't figure out how to set <a href="http://voip.ms" target="_blank">voip.ms</a> to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way?<u></u><u></u></p>
<div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">Thanks a bunch!<u></u><u></u></p><div><p class="MsoNormal">On Mon, Sep 12, 2011 at 11:18 PM, naren <<a href="mailto:naren.salem@gmail.com" target="_blank">naren.salem@gmail.com</a>> wrote:<u></u><u></u></p>
<p class="MsoNormal">I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk.<u></u><u></u></p>
<div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with <a href="http://voip.ms" target="_blank">voip.ms</a>. <u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :)<u></u><u></u></p></div><div>
<p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thanks<u></u><u></u></p></div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Mon, Sep 12, 2011 at 11:09 AM, John Novack <<a href="mailto:jnovack@stromberg-carlson.org" target="_blank">jnovack@stromberg-carlson.org</a>> wrote:<u></u><u></u></p>
<div><p class="MsoNormal">Never have had a problem with their IAX service.<br><br>And ( for now ) a little hedge against the hackers.<br><br>Since Asterisk is involved, why not use IAX anyway?<br><span style="color:#888888"><br>
<br>John Novack</span><u></u><u></u></p><div><div><p class="MsoNormal"><br><br><br>naren wrote: <u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I also found this... seems like <a href="http://voip.ms" target="_blank">voip.ms</a> outbound is broken for now!<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><a href="http://pbxinaflash.com/forum/showthread.php?t=10735" target="_blank">http://pbxinaflash.com/forum/showthread.php?t=10735</a><u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Sun, Sep 11, 2011 at 10:34 PM, naren <<a href="mailto:naren.salem@gmail.com" target="_blank">naren.salem@gmail.com</a>> wrote:<u></u><u></u></p>
<p class="MsoNormal">Hi, <u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I am trying to set up my asterisk 1.8.5 with <a href="http://voip.ms" target="_blank">voip.ms</a>. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website?<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert.<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I would really appreciate it if you could post the relevant section of your sip.conf for me.<u></u><u></u></p></div><div><p class="MsoNormal">
<u></u> <u></u></p></div><div><p class="MsoNormal">Thanks!<u></u><u></u></p></div><div><p class="MsoNormal">Naren<u></u><u></u></p></div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p>
<div><p class="MsoNormal">On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards <<a href="mailto:asterisk.org@sedwards.com" target="_blank">asterisk.org@sedwards.com</a>> wrote:<u></u><u></u></p><div><p class="MsoNormal" style="margin-bottom:12.0pt">
On Thu, 9 Jun 2011, John Novack wrote:<u></u><u></u></p><p class="MsoNormal">I use <a href="http://voip.ms" target="_blank">voip.ms</a> and have no issues using IAX and Asterisk 1.4.xx<u></u><u></u></p><p class="MsoNormal">
<u></u> <u></u></p></div><p class="MsoNormal">'slam-dunk.' <u></u><u></u></p><div><blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in"><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u> <u></u></p><p class="MsoNormal">Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall<u></u><u></u></p></blockquote><p class="MsoNormal" style="margin-bottom:12.0pt"><br>a<u></u><u></u></p><p class="MsoNormal">
Their on line config samples just work!<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal">is <u></u><u></u></p><div><blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><p class="MsoNormal">Suggest you check your firewall and your configs, and above all post some more information<u></u><u></u></p></blockquote><p class="MsoNormal">
<u></u> <u></u></p></div><p class="MsoNormal">IAX <u></u><u></u></p><div><blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in"><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u> <u></u></p><p class="MsoNormal">If you really want to upset some, top post as I have just done!<u></u><u></u></p></blockquote><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal">Agreed. <u></u><u></u></p>
<div><blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in"><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><p class="MsoNormal">The real issue is communication, top bottom or in the middle<u></u><u></u></p>
</blockquote><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal">Sometimes, it's just about being considerate to 'the next guy.'<br><span style="color:#888888"><br>-- <br>Thanks in advance,<br>-------------------------------------------------------------------------<br>
Steve Edwards <a href="mailto:sedwards@sedwards.com" target="_blank">sedwards@sedwards.com</a> Voice: <a href="tel:%2B1-760-468-3867" target="_blank">+1-760-468-3867</a> PST<br>Newline Fax: <a href="tel:%2B1-760-731-3000" target="_blank">+1-760-731-3000</a></span> <u></u><u></u></p>
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