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<div><br>
</div><div>x SIP/IAX - [main account] IAX2/100000 <- with my account number</div><div>x SIP URI - <a href="http://SIP:mysipid@myuri.com:5060">SIP:mysipid@myuri.com:5060</a></div><div>x System - Hangup</div><div><br></div>
<div>There are several other options but they are not selectable for me because I have not set up to use them.</div><div><br></div><div>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. </div>
<div><br></div><div>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. </div>
<div><br></div><div>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.</div>
<div><br></div><div>Thanks. <br><br><div class="gmail_quote">On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel <span dir="ltr"><<a href="mailto:daibel@pervasivetelecom.com">daibel@pervasivetelecom.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">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.<br>
<div><div></div><div class="h5"><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" value="+17604683867">+1-760-468-3867</a> PST<br>
>> Newline Fax: <a href="tel:%2B1-760-731-3000" value="+17607313000">+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>
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>><br>
>> asterisk-users mailing list<br>
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>><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>
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><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>
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><br>
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><br>
<br>
<br>
<br>
--<br>
</div></div>+++++++++++++++++++++++++++++++++++++++++<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" value="+16033673512">(603)367.3512</a><br>
<a href="tel:%28603%29367.9942" value="+16033679942">(603)367.9942</a><br>
<a href="tel:%28401%29862.4203" value="+14018624203">(401)862.4203</a> (c)<br>
<br>
<a href="mailto:daibel@pervasivetelcom.com">daibel@pervasivetelcom.com</a><br>
<div><div></div><div class="h5"><br>
--<br>
_____________________________________________________________________<br>
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New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
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</div></div></blockquote></div><br></div>