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I have verified all of my steps based on the wiki. When I make an
outbound call (or any call that would affect NAT), I hear the ring for
a second than it disappears. The console shows:<br>
<br>
2008-09-08 17:06:54 [INFO] mod_sofia.c:1085 sofia_receive_message()
Asked to send early media by <a class="moz-txt-link-abbreviated" href="mailto:sofia/internal/1006@75.101.142.208">sofia/internal/1006@75.101.142.208</a><br>
2008-09-08 17:06:54 [INFO] mod_sofia.c:1126 sofia_receive_message()
Ring SDP:<br>
v=0<br>
o=FreeSWITCH 1220867834 1220867835 IN IP4 10.251.38.32<br>
s=FreeSWITCH<br>
c=IN IP4 10.251.38.32<br>
t=0 0<br>
a=sendrecv<br>
m=audio 25780 RTP/AVP 0 101 13<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:30<br>
<br>
2008-09-08 17:06:54 [NOTICE] mod_sofia.c:1129 sofia_receive_message()
Pre-Answer <a class="moz-txt-link-abbreviated" href="mailto:sofia/internal/1006@75.101.142.208">sofia/internal/1006@75.101.142.208</a>!<br>
2008-09-08 17:07:11 [NOTICE] sofia.c:2573 sofia_handle_sip_i_state()
Hangup <a class="moz-txt-link-abbreviated" href="mailto:sofia/internal/1006@75.101.142.208">sofia/internal/1006@75.101.142.208</a> [CS_EXECUTE]
[ORIGINATOR_CANCEL]<br>
2008-09-08 17:07:11 [NOTICE] switch_ivr_bridge.c:384
audio_bridge_on_exchange_media() Hangup
<a class="moz-txt-link-abbreviated" href="mailto:sofia/external/19169621931@sjc-primary.voicepulse.com">sofia/external/19169621931@sjc-primary.voicepulse.com</a> [CS_RESET]
[NORMAL_CLEARING]<br>
<br>
I used the Quick and Dirty Install .... I believe I'm close and just
missing something. Any other thoughts?<br>
<br>
Thanks all for your help<br>
<br>
Damon<br>
<br>
Chris Danielson wrote:
<blockquote cite="mid:48C549A7.7040007@maxpowersoft.com" type="cite">
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
Updated the wiki <a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Amazon_ec2">http://wiki.freeswitch.org/wiki/Amazon_ec2</a>
with Novak's notes.<br>
<br>
Kind Regards,<br>
Chris<br>
<br>
Novak Joe wrote:
<blockquote
cite="mid:2e45e80809080837y62861ff8o60656fb6c460f242@mail.gmail.com"
type="cite">
<pre wrap="">Hi,
When I first tried setting up freeswitch on ec2 I attempted to do
this through the quick and dirty wiki:
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install">http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install</a>
and the ec2 wiki
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://wiki.freeswitch.org/wiki/Amazon_ec2">http://wiki.freeswitch.org/wiki/Amazon_ec2</a>
This didn't work out for me. You also need to take a careful look at
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall">http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Firewall</a>
then, for sip, setup a security group with the following ports enabled
(easiest to do with elasticfox
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://developer.amazonwebservices.com/connect/entry.jspa?externalID=609">http://developer.amazonwebservices.com/connect/entry.jspa?externalID=609</a>
):
udp 16384:32768
udp 4569
udp 5060
tcp 5060
udp 5080
tcp 5080
tcp 8000
udp 8000
make sure you make the security group, and apply it before you boot
the instance.
after this, the quick and dirty install guid worked just fine for
me. you'll just need to tweak the default dialplan
to your needs.
cheers
Hi Brian,
Did you have any luck with the EC2 instances. I tried a few things, but
I just cant get RTP to run through it. PLease let me know.
Best
Damon
Brian West wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Let me launch mine in 32bit and see what I can do over the weekend...
I had this working without a problem. ;/
/b
On Sep 6, 2008, at 12:09 AM, Damon Brown wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Ive Tried the following with no success:
internal.xml
<param name="ext-rtp-ip" value="75.101.142.208"/>
<param name="ext-sip-ip" value="75.101.142.208"/>
Ive also tried just changing the vars.xml file. I also tried
forwarding the incoming rtp connections to my test pc. all with no
audio success. I am sure im doing something wrong I jsut cant find
it.
I found another suggestion on the wiki of creating a "double nat"
that listens on 5090. That didnt change anything either, here is
that information:
<settings>
</pre>
</blockquote>
</blockquote>
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