Ok... this is probably a dumb question but I can't figure out how to set <a href="http://voip.ms">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?<div>
<br></div><div>Thanks a bunch!<br><br><div class="gmail_quote">On Mon, Sep 12, 2011 at 11:18 PM, naren <span dir="ltr"><<a href="mailto:naren.salem@gmail.com">naren.salem@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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.<div>
<br></div><div>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>. </div>
<div><br></div><div>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 :)</div><div><br></div><div>Thanks</div><div><div></div><div class="h5"><div><br><br><div class="gmail_quote">
On Mon, Sep 12, 2011 at 11:09 AM, John Novack <span dir="ltr"><<a href="mailto:jnovack@stromberg-carlson.org" target="_blank">jnovack@stromberg-carlson.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
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><font color="#888888">
<br>
<br>
John Novack</font><div><div></div><div><br>
<br>
<br>
naren wrote:
<blockquote type="cite">
<div><br>
</div>
<div>I also found this... seems like <a href="http://voip.ms" target="_blank">voip.ms</a> outbound is broken for now!</div>
<div><br>
</div>
<div><a href="http://pbxinaflash.com/forum/showthread.php?t=10735" target="_blank">http://pbxinaflash.com/forum/showthread.php?t=10735</a></div>
<div><br>
</div>
<br>
<br>
<div class="gmail_quote">On Sun, Sep 11, 2011 at 10:34 PM, naren <span dir="ltr"><<a href="mailto:naren.salem@gmail.com" target="_blank">naren.salem@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,
<div><br>
</div>
<div>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?</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>I would really appreciate it if you could post the
relevant section of your sip.conf for me.</div>
<div><br>
</div>
<div>Thanks!</div>
<div>Naren</div>
<div>
<div>
<div><br>
<br>
<div class="gmail_quote">
On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards <span dir="ltr"><<a href="mailto:asterisk.org@sedwards.com" target="_blank">asterisk.org@sedwards.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>On Thu, 9 Jun 2011, John Novack wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I use <a href="http://voip.ms" target="_blank">voip.ms</a>
and have no issues using IAX and Asterisk 1.4.xx<br>
</blockquote>
<br>
</div>
'slam-dunk.'
<div><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Though they suggest SIP, I chose IAX and have
4569 UDP open in my firewall<br>
</blockquote>
<br>
a<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Their on line config samples just work!<br>
</blockquote>
<br>
</div>
is
<div><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Suggest you check your firewall and your
configs, and above all post some more
information<br>
</blockquote>
<br>
</div>
IAX
<div><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
If you really want to upset some, top post as I
have just done!<br>
</blockquote>
<br>
</div>
Agreed.
<div><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
The real issue is communication, top bottom or
in the middle<br>
</blockquote>
<br>
</div>
Sometimes, it's just about being considerate to 'the
next guy.'<br>
<font 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" value="+17604683867" target="_blank">+1-760-468-3867</a>
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Newline
Fax: <a href="tel:%2B1-760-731-3000" value="+17607313000" target="_blank">+1-760-731-3000</a></font>
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<div><br>
<br>
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</blockquote>
</div>
<br>
</div>
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</blockquote>
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</div></div><div><pre cols="10000">--
Dog is my Co-pilot</pre>
</div></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>