<div dir="ltr">Hi,<div><br></div><div>Is it only during voicemal? If so, Try this:</div><div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Variable_record_waste_resources">http://wiki.freeswitch.org/wiki/Variable_record_waste_resources</a><br>
</div><div><br></div><div>Otherwise turn on sip trace with "sofia global siptrace on" and "console loglevel debug" and post it.</div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Thu, Jul 18, 2013 at 10:54 AM, Paul <span dir="ltr"><<a href="mailto:pasha@prosperity4ever.com" target="_blank">pasha@prosperity4ever.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi guys,<br>
<br>
After hours of googling and trying every different config options I need<br>
some help!<br>
<br>
My scenario is as follows:<br>
<br>
Remote Network 1: <a href="http://192.168.1.0/24" target="_blank">192.168.1.0/24</a> (10.8.0.X/32 via openVPN tun)<br>
Remote Network 2: <a href="http://192.168.5.0/24" target="_blank">192.168.5.0/24</a> (10.9.0.X/32 via openVPN tun)<br>
<br>
Main Network (where Freeswitch resides): <a href="http://10.0.0.0/24" target="_blank">10.0.0.0/24</a><br>
<br>
My sip gateway/proxy: 10.0.0.40 (kamailio)<br>
Freeswitch: 10.0.0.34<br>
<br>
I can fully use the VPN from both networks (ssh, web gui control, etc)<br>
<br>
phones can register, however here are my problems:<br>
<br>
When calling incoming DID none of the extensions ever get ring, it skips<br>
to voicemail... when leaving a message voicemail cuts of after 30-31<br>
seconds exactly. I am wondering if freeswitch isn't detecting my client<br>
sending audio and interprets it as silence and therefore hangs up. The<br>
voicemail message itsel does contain everything said right up to the hang<br>
up.<br>
<br>
Sometimes I can call between extensions if they are on the same side of<br>
the VPN subnet (say 101 calling 101 and they both are coming from<br>
<a href="http://192.168.1.0/24" target="_blank">192.168.1.0/24</a> via <a href="http://10.8.0.0/24" target="_blank">10.8.0.0/24</a>)<br>
If 100 is on 10.8.0/24 and 101 is on <a href="http://10.9.0.0/24" target="_blank">10.9.0.0/24</a> freeswitch just says user<br>
is unavailable.<br>
<br>
I'm suspecting this is a NAT issue, all the reading I have done talks<br>
about "external sip and rtp ip" in my case my trunk hooks up from private<br>
ip to private ip, it's the kamailio that's supposed to do the NAT (which I<br>
have verified works fine, I can place outgoing calls through it from other<br>
PBXs no problem with two way audio and working as it should be).<br>
<br>
I'm not sure if there is a way to enable nat or not with freeswitch. One<br>
thing I have done (as I've found this worked for some people) is set all<br>
references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my<br>
freeswitch eth0). There are no other adapters on the box.<br>
<br>
I am hoping that as this is my first time setting up freeswitch that one<br>
of you freeswitch experts can point me in the right direction :)<br>
<br>
Thanks in advance guys!<br>
<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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