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<DIV><FONT size=2>I had same problem from this provider without to explain
why.</FONT></DIV>
<DIV><FONT size=2>One day it works, another day it doesn't, their support is
crap...</FONT></DIV>
<DIV> </DIV>
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style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=anthony.minessale@gmail.com
href="mailto:anthony.minessale@gmail.com">Anthony Minessale</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Wednesday, July 07, 2010 2:37
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] FS 1.0.6
on EC2 - no audio on outgoing calls</DIV>
<DIV><BR></DIV>not really, not with so little information.
<DIV><BR><BR>
<DIV class=gmail_quote>On Wed, Jul 7, 2010 at 1:30 PM, paul gore <SPAN
dir=ltr><<A
href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</A>></SPAN>
wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>Firewall is configured according to the wiki, I also tried
to open all<BR>udp ports, issue persists.<BR>Actually the problem became
more complex - outgoing calls don't work<BR>with one particular termination
provider, <A href="http://siptraffic.com" target=_blank>siptraffic.com</A> ,
any ideas<BR>why?<BR>Outgoing calls also don't work when originating a call
via js script<BR>or via FS api. Any clues on that one?<BR>
<DIV>
<DIV></DIV>
<DIV class=h5><BR>On 7/6/10, paul gore <<A
href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</A>>
wrote:<BR>> Hi there,<BR>> I am experimenting with FS on EC2, I like
results, but stuck on weird audio<BR>> issue - I followed FreeSwitch EC2
wiki article and modified internal<BR>> profile<BR>> and vars.xml
accordingly, but unfortunately still cannot get it working.<BR>> Incoming
and outgoing calls made using a SIP phone to FS extensions work<BR>> just
fine. As well as calls to FS from PSTN. But calls to PSTN via
gateways<BR>> result in no audio at all, no ring, nothing, SIP signaling
goes through OK.<BR>> Sofia status profile shows correct values for
Ext-RTP-IP for both profiles<BR>> -<BR>> my static public IP, RTP-IP
shows local IP.<BR>> Any thoughts on that? Anybody can share working
profile configuration may<BR>> be?<BR>> Please help, I really need to
get this going.<BR>><BR>> Thanks.<BR>><BR><BR></DIV></DIV>
<DIV>
<DIV></DIV>
<DIV
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clear=all><BR>-- <BR>Anthony Minessale II<BR><BR>FreeSWITCH <A
href="http://www.freeswitch.org/">http://www.freeswitch.org/</A><BR>ClueCon <A
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#freeswitch<BR><BR>FreeSWITCH Developer Conference<BR><A
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