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Hi,<br>
<br>
If your cluster's virtual IP is using ip aliasing (eg eth0:0), i
think your problem come from UDP flows, they are, in opposition to
TCP flows, unconnected, so the IP stack take the shortest
route/interface to send them, wich is when this is the default
route, the real interface and not the aliased.<br>
<br>
For exemple if you have eth0 the real, eth0:0 the virtual, you can
try to add in your failback/failover cluster script something like
this :<br>
<br>
# when the virtual ip come up<br>
ip r a "SIP_PROVIDER_IP" via "GATEWAY_IP" dev eth0:0<br>
<br>
# when the virtual ip come down, maybe facultative because the route
is deleted when the interface fall down<br>
ip r d "SIP_PROVIDER_IP" via "GATEWAY_IP" dev eth0:0<br>
<br>
Regards,<br>
<br>
Cédric<br>
<br>
<br>
Le 15/06/11 19:01, <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a> a écrit :
<blockquote
cite="mid:mailman.11.1308157272.11473.asterisk-users@lists.digium.com"
type="cite">
<pre wrap="">Date: Wed, 15 Jun 2011 17:28:36 +0200
From: <a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:rosenberger@taoweb.at">rosenberger@taoweb.at</a>
Subject: [asterisk-users] connecting to SIP Provider with virtual IP
        from        pacemaker cluster
To: <a moz-do-not-send="true" class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
Message-ID: <a moz-do-not-send="true" class="moz-txt-link-rfc2396E" href="mailto:d265002689c5927864dc1da6ae1f1e0b@taoweb.at"><d265002689c5927864dc1da6ae1f1e0b@taoweb.at></a>
Content-Type: text/plain; charset=UTF-8; format=flowed
Hello !
i am new to this list and asterisk.
I run asterisk 1.4 on a OpenSuSE 11.4. My SIP Provider needs my IP to
connect the local area number to my IP and also for there firewall.
I plan to run asterisk in a pacemaker cluster that is not the problem
and works.
My problem is the virtual IP from the cluster and the connection to the
SIP Provider.
My Server has 2 NIC's one intern to register the VOIP Phone's and one
external to register to the SIP Provider.
If i run asterisk with a virtuail IP configuration i get problems to
connect to the SIP Provider
because the ip route uses the real Hardware NIC ip.
OK this i can solve with setting a sorce IP with ip route. That works
with ssh an other but not with asterisk.
tcpdump always show the the ip from the real Hardware device.
A bindaddr in the SIP conf is first not possible because i had to
connect from internal and external.
But when i try a bindaddr on the virtual device tcpdump shows the same
result that the connection comes
from the real IP.
maybe some had the same problem an can give me a hint.
BR/Torsten</pre>
</blockquote>
<br>
<br>
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<div><b><font color="#003f80" face="Arial" size="2">Cédric
Lemarchand<br>
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