<div dir="ltr">I'm guessing that OpenSIPS is behind nat with FreeSWITCH?</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Aug 20, 2014 at 5:59 AM, Matt Broad <span dir="ltr"><<a href="mailto:matt@inveroak.com" target="_blank">matt@inveroak.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
Hi Brian,<br>
<br>
it seems that this was not the issue.<br>
<br>
Some more details:<br>
We have an opensips server using the load_balancer module to route
the calls to the freeswitch box.
(<a href="https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS" target="_blank">https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS</a>)<br>
When I route the calls directly to the freeswitch's external IP, the
calls connect fine and have audio.<br>
However when I route the calls to the opensips server first, the
call will connect but without audio. <br>
<br>
The strange thing is Opensips uses Freeswitch's internal address
(from the load_balancer table) to forward the INVITE packet, yet
when the ACK packet comes in from the provider opensips tries to
forward it to the FS external IP address rather than the internal
IP.<br>
<br>
I understand this is probably more an opensips question, so will
understand if I should post this to their mailing list, but any tips
would be appreciated. <br>
<br>
<br>
thanks<span class="HOEnZb"><font color="#888888"><br>
Matt</font></span><div><div class="h5"><br>
<br>
<br>
<div>On 19/08/2014 17:25, Matt Broad wrote:<br>
</div>
<blockquote type="cite">
thanks brian, will give it a go and see what happens.<br>
<br>
thanks<br>
Matt<br>
<div>On 19/08/2014 17:06, Brian West
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Can you try prefixing the ext-rtp-ip and
ext-sip-ip with autonat:0.0.0.0 and make sure
local-network-acl is set to rfc1918.auto?</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Tue, Aug 19, 2014 at 9:20 AM, Matt
Broad <span dir="ltr"><<a href="mailto:matt@inveroak.com" target="_blank">matt@inveroak.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
does it make any difference if my internal IP address is
being sent in<br>
the SDP connection information? I have 2 sip providers
where 1 I can<br>
bridge calls and receive audio and 1 that I can bridge
calls but get no<br>
audio.<br>
<br>
Even with the below settings the SDP message body contains
c=IN IP4<br>
192.168.1.1.<br>
<br>
I have vars set as:<br>
<X-PRE-PROCESS cmd="set"
data="external_rtp_ip=my.ip.goes.here"/><br>
<X-PRE-PROCESS cmd="set"
data="external_sip_ip=my.ip.goes.here"/><br>
<br>
I have setup the external profile as follows:<br>
<param name="ext-rtp-ip"
value="$${external_rtp_ip}"/><br>
<param name="ext-sip-ip"
value="$${external_sip_ip}"/><br>
<br>
running sofia status profile external gives (replace
0.0.0.0 with the<br>
real external IP declared in vars):<br>
=================================================================================================<br>
Name external<br>
Domain Name N/A<br>
Auto-NAT false<br>
DBName sofia_reg_external<br>
Pres Hosts<br>
Dialplan XML<br>
Context public<br>
Challenge Realm auto_to<br>
RTP-IP 192.168.1.1<br>
Ext-RTP-IP 0.0.0.0<br>
SIP-IP 192.168.1.1<br>
Ext-SIP-IP 0.0.0.0<br>
URL <a href="http://sip:mod_sofia@0.0.0.0:5080" target="_blank">sip:mod_sofia@0.0.0.0:5080</a><br>
BIND-URL <a href="mailto:sip:mod_sofia@0.0.0.0:5080;maddr=192.168.1.1;transport=udp,tcp" target="_blank">sip:mod_sofia@0.0.0.0:5080;maddr=192.168.1.1;transport=udp,tcp</a><br>
WS-BIND-URL <a href="mailto:sip:mod_sofia@192.168.1.1:5086;transport=ws" target="_blank">sip:mod_sofia@192.168.1.1:5086;transport=ws</a><br>
HOLD-MUSIC local_stream://moh<br>
OUTBOUND-PROXY N/A<br>
CODECS IN G722,PCMU,PCMA,GSM<br>
CODECS OUT PCMU,PCMA,GSM<br>
TEL-EVENT 101<br>
DTMF-MODE rfc2833<br>
CNG 13<br>
SESSION-TO 0<br>
MAX-DIALOG 0<br>
NOMEDIA false<br>
LATE-NEG true<br>
PROXY-MEDIA false<br>
ZRTP-PASSTHRU true<br>
AGGRESSIVENAT false<br>
CALLS-IN 8<br>
FAILED-CALLS-IN 1<br>
CALLS-OUT 8<br>
FAILED-CALLS-OUT 1<br>
REGISTRATIONS 0<br>
<br>
<br>
<br>
<br>
thanks<br>
Matt<br>
<br>
<br>
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-- <br>
<div dir="ltr">
<p><font face="courier new, monospace"><b><i><font size="4">Brian
West</font></i></b><br>
<span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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