<div><br></div><div>I also found this... seems like <a href="http://voip.ms">voip.ms</a> outbound is broken for now!</div><div><br></div><div><a href="http://pbxinaflash.com/forum/showthread.php?t=10735">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">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><div class="h5"><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>
------------------------------<u></u>------------------------------<u></u>-------------<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> PST<br>
Newline Fax: <a href="tel:%2B1-760-731-3000" value="+17607313000" target="_blank">+1-760-731-3000</a></font><div><div></div><div><br>
<br>
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