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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'>That’s what this part of extensions.conf should do:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN style='font-size:10.0pt;font-family:"Courier New"'>; inbound context example for your DID numbers, do not add the number 1 in front<o:p></o:p></span></p><p class=MsoNormal><span lang=EN style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN style='font-size:10.0pt;font-family:"Courier New"'>[voipms-inbound]<o:p></o:p></span></p><p class=MsoNormal><span lang=EN style='font-size:10.0pt;font-family:"Courier New"'>exten => 7863643011,1,Answer() ;your DID<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><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 4:09 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>Yup, that part I got. What I am not clear about is how to set up the DID to go to my URI. When I select "manage DIDs" and click on the one I want to change, I see the following options for routing the DID<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>x SIP/IAX - [main account] IAX2/100000 <- with my account number<o:p></o:p></p></div><div><p class=MsoNormal>x SIP URI - <a href="http://SIP:mysipid@myuri.com:5060">SIP:mysipid@myuri.com:5060</a><o:p></o:p></p></div><div><p class=MsoNormal>x System - Hangup<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>There are several other options but they are not selectable for me because I have not set up to use them.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I used to have the routing set to SIP URI where I was able to specify my URI where the call was routed to. But with the SIP/IAX option I do not have that ability. <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I am missing something fundamental here. My asterisk has the iax.conf and extensions.conf entries ready to receive calls from <a href="http://voip.ms">voip.ms</a>, but I don't know how to tel <a href="http://voip.ms">voip.ms</a> to send the calls to my asterisk with the IAX protocol. <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I understand this is probably a question for the <a href="http://voip.ms">voip.ms</a> folks, but since a couple of people mentioned earlier that they were rocking with IAX, I thought it would be an easy question for them to point me in the right direction.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'>Thanks. <o:p></o:p></p><div><p class=MsoNormal>On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel <<a href="mailto:daibel@pervasivetelecom.com">daibel@pervasivetelecom.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>I was lurking in this conversation and I went to look more carefully<br>at the <a href="http://voip.ms" target="_blank">voip.ms</a> site. I found sample files at<br><a href="http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29" target="_blank">http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29</a><br><br>Hope that helps.<o:p></o:p></p><div><div><p class=MsoNormal><br><br>On Tue, Sep 13, 2011 at 3:59 PM, naren <<a href="mailto:naren.salem@gmail.com">naren.salem@gmail.com</a>> wrote:<br>> I see the section you are talking about. It is on the home page if I am not<br>> logged in. I see the Authentication section and the text "IAX/SIP<br>> registration", but it doesn't seem to be a link. I am not sure how I can<br>> find the page that has the details about the IAX/SIP registration. I see in<br>> the wiki there is a page that has the configuration info for iax.conf and<br>> extensions.conf.<br>> Thanks for your help.<br>> naren<br>><br>> On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>> wrote:<br>>><br>>> Did you read the “IAX/SIP registration” section (under Authentication) on<br>>> <a href="http://voip.ms" target="_blank">voip.ms</a>?<br>>><br>>><br>>><br>>> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>>> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of naren<br>>> Sent: Tuesday, September 13, 2011 2:22 PM<br>>> To: John Novack<br>>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion<br>>> Subject: Re: [asterisk-users] Question about <a href="http://voip.ms" target="_blank">voip.ms</a> service.<br>>><br>>><br>>><br>>> Ok... this is probably a dumb question but I can't figure out how to set<br>>> <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<br>>> pointed it to my asterisk installation, but with IAX I don't have that<br>>> option. Is that supposed to work some other way?<br>>><br>>><br>>><br>>> Thanks a bunch!<br>>><br>>> On Mon, Sep 12, 2011 at 11:18 PM, naren <<a href="mailto:naren.salem@gmail.com">naren.salem@gmail.com</a>> wrote:<br>>><br>>> I am novice with Asterisk, I had to piece together a lot of bits of info<br>>> from lots of internet searches to get my very basic setup working. I<br>>> probably shouldn't say that because it seems like Nat is not a very basic<br>>> setup with Asterisk.<br>>><br>>><br>>><br>>> The reason for wanting to stay with SIP is because I have my setup working<br>>> with that protocol with an incoming and an outgoing line. I just wanted to<br>>> add a second outgoing with <a href="http://voip.ms" target="_blank">voip.ms</a>.<br>>><br>>><br>>><br>>> But, I have come so far, so well why not... I will give IAX a shot, and<br>>> see what traps I need to wade through :)<br>>><br>>><br>>><br>>> Thanks<br>>><br>>><br>>><br>>> On Mon, Sep 12, 2011 at 11:09 AM, John Novack<br>>> <<a href="mailto:jnovack@stromberg-carlson.org">jnovack@stromberg-carlson.org</a>> wrote:<br>>><br>>> 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>>><br>>><br>>> John Novack<br>>><br>>><br>>> naren wrote:<br>>><br>>><br>>><br>>> I also found this... seems like <a href="http://voip.ms" target="_blank">voip.ms</a> outbound is broken for now!<br>>><br>>><br>>><br>>> <a href="http://pbxinaflash.com/forum/showthread.php?t=10735" target="_blank">http://pbxinaflash.com/forum/showthread.php?t=10735</a><br>>><br>>><br>>><br>>><br>>><br>>> On Sun, Sep 11, 2011 at 10:34 PM, naren <<a href="mailto:naren.salem@gmail.com">naren.salem@gmail.com</a>> wrote:<br>>><br>>> Hi,<br>>><br>>><br>>><br>>> 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<br>>> with the incoming, but my outgoing is not working. If at all possible, I<br>>> would like to stick with SIP. Since the original poster (Glen) had mentioned<br>>> that he had gotten outgoing working, I was wondering if you would be kind<br>>> enough to post some thoughts on that. Were you able to get it working with<br>>> just the default example sip.conf / extensions.conf settings that they have<br>>> on their website?<br>>><br>>><br>>><br>>> I have pretty much the same settings. When I dial out, the destination<br>>> rings, but I can't hear a ringback tone from on the source side ( I am using<br>>> a PAP2T router with a phone). I have set up outgoing with actionvoip before<br>>> and that is working fine, so I am thinking my router settings for my ports<br>>> are correct - but I am no expert.<br>>><br>>><br>>><br>>> I would really appreciate it if you could post the relevant section of<br>>> your sip.conf for me.<br>>><br>>><br>>><br>>> Thanks!<br>>><br>>> Naren<br>>><br>>><br>>><br>>> On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards <<a href="mailto:asterisk.org@sedwards.com">asterisk.org@sedwards.com</a>><br>>> wrote:<br>>><br>>> On Thu, 9 Jun 2011, John Novack wrote:<br>>><br>>> I use <a href="http://voip.ms" target="_blank">voip.ms</a> and have no issues using IAX and Asterisk 1.4.xx<br>>><br>>><br>>><br>>> 'slam-dunk.'<br>>><br>>><br>>><br>>> Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall<br>>><br>>> a<br>>><br>>> Their on line config samples just work!<br>>><br>>><br>>><br>>> is<br>>><br>>><br>>><br>>> Suggest you check your firewall and your configs, and above all post some<br>>> more information<br>>><br>>><br>>><br>>> IAX<br>>><br>>><br>>><br>>> If you really want to upset some, top post as I have just done!<br>>><br>>><br>>><br>>> Agreed.<br>>><br>>><br>>><br>>> The real issue is communication, top bottom or in the middle<br>>><br>>><br>>><br>>> Sometimes, it's just about being considerate to 'the next guy.'<br>>><br>>> --<br>>> Thanks in advance,<br>>> -------------------------------------------------------------------------<br>>> Steve Edwards <a href="mailto:sedwards@sedwards.com">sedwards@sedwards.com</a> Voice: <a href="tel:%2B1-760-468-3867">+1-760-468-3867</a> PST<br>>> Newline Fax: <a href="tel:%2B1-760-731-3000">+1-760-731-3000</a><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><br>>><br>>><br>>><br>>> --<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>>><br>>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>>><br>>> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>>><br>>><br>>><br>>> asterisk-users mailing list<br>>><br>>> To UNSUBSCRIBE or update options visit:<br>>><br>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>>><br>>><br>>><br>>> --<br>>><br>>><br>>><br>>> Dog is my Co-pilot<br>>><br>>><br>>><br>>><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><br>><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><br>><br><br><br><br>--<o:p></o:p></p></div></div><p class=MsoNormal>+++++++++++++++++++++++++++++++++++++++++<br>Dave Aibel<br><br>President & CEO<br>Pervasive Telecommunications, Inc.<br><br>email: <a href="mailto:daibel@pervasivetelecom.com">daibel@pervasivetelecom.com</a><br><br><a href="tel:%28603%29367.3512">(603)367.3512</a><br><a href="tel:%28603%29367.9942">(603)367.9942</a><br><a href="tel:%28401%29862.4203">(401)862.4203</a> (c)<br><br><a href="mailto:daibel@pervasivetelcom.com">daibel@pervasivetelcom.com</a><o:p></o:p></p><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></div></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>