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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I see what you mean. Maybe if you call their support they can tell you what you need to know. If not, voicepulse is a pretty good provider.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>naren<br><b>Sent:</b> Tuesday, September 13, 2011 2:59 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Question about voip.ms service.<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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. <o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thanks for your help.<o:p></o:p></p></div><div><p class=MsoNormal>naren<o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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>? </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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.</span><o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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?<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'>Thanks a bunch!<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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:<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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.<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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>. <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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 :)<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Thanks<o:p></o:p></p></div><div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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:<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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><o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><br><br><br>naren wrote: <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I also found this... seems like <a href="http://voip.ms" target="_blank">voip.ms</a> outbound is broken for now!<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><a href="http://pbxinaflash.com/forum/showthread.php?t=10735" target="_blank">http://pbxinaflash.com/forum/showthread.php?t=10735</a><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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:<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Hi, <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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?<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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.<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I would really appreciate it if you could post the relevant section of your sip.conf for me.<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Thanks!<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Naren<o:p></o:p></p></div><div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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:<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'>On Thu, 9 Jun 2011, John Novack wrote:<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I use <a href="http://voip.ms" target="_blank">voip.ms</a> and have no issues using IAX and Asterisk 1.4.xx<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>'slam-dunk.' <o:p></o:p></p><div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall<o:p></o:p></p></blockquote><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><br>a<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Their on line config samples just work!<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>is <o:p></o:p></p><div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Suggest you check your firewall and your configs, and above all post some more information<o:p></o:p></p></blockquote><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>IAX <o:p></o:p></p><div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>If you really want to upset some, top post as I have just done!<o:p></o:p></p></blockquote><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Agreed. <o:p></o:p></p><div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-bottom:5.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>The real issue is communication, top bottom or in the middle<o:p></o:p></p></blockquote><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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> <o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><o:p> </o:p></p><pre>--<o:p></o:p></pre><pre>_____________________________________________________________________<o:p></o:p></pre><pre>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<o:p></o:p></pre><pre>New to Asterisk? Join us for a live introductory webinar every Thurs:<o:p></o:p></pre><pre> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><o:p></o:p></pre><pre> <o:p></o:p></pre><pre>asterisk-users mailing list<o:p></o:p></pre><pre>To UNSUBSCRIBE or update options visit:<o:p></o:p></pre><pre> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></pre><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div><div><pre>-- <o:p></o:p></pre><pre> <o:p></o:p></pre><pre>Dog is my Co-pilot<o:p></o:p></pre></div></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div></div></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div></div></div></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>