[asterisk-users] Use SRV for failover proxy

2013-09-06 Thread Dominique Haeber
Hi all,

is it possible that asterisk uses two proxies with SRV?
The enddevices are registered on one of the two Proxies (Kamailio).
The two proxies communicate with each other. 
And asterisk can choose one of this proxies with SRV.

asterisk
|   \
|\
Proxy1Proxy2


I have tries to solve this problem with two trunks for this proxies
and Dial(...@proxytrunk) but on this way the properties of the
enddevices are lost and the properties from trunk are taken.

So i'd like to use SRV, if that is possible.

Very thanks
Dominique


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Re: [asterisk-users] Use SRV for failover proxy

2013-09-06 Thread Gareth Blades

On 06/09/13 09:42, Dominique Haeber wrote:

Hi all,

is it possible that asterisk uses two proxies with SRV?
The enddevices are registered on one of the two Proxies (Kamailio).
The two proxies communicate with each other.
And asterisk can choose one of this proxies with SRV.

asterisk
|   \
|\
Proxy1Proxy2


I have tries to solve this problem with two trunks for this proxies
and Dial(...@proxytrunk) but on this way the properties of the
enddevices are lost and the properties from trunk are taken.

So i'd like to use SRV, if that is possible.

Very thanks
Dominique
No asterisk will always use the first SRV record and wont load balance 
or switch to a backup if its not reachable.


What we do is have each endpoint defined in sip.conf with qualify=yes 
and then in the dialplan use the ${SIPPEER(x)} variable to pull out the 
status of each peer and pass it into an AGI application to perform the 
load balancing etc...


If you are happy with wone being a primary and one being a backup then 
if you have qualify=yes set for both you could just dial using the first 
one and then an execif hangupcause=20 then try dialing the backup.


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Re: [asterisk-users] Use SRV for failover proxy

2013-09-06 Thread Dominique Haeber
Gareth Blades mailinglist+aster...@dns99.co.uk schrieb am Fre, 06. Sep 13:21:
 No asterisk will always use the first SRV record and wont load
 balance or switch to a backup if its not reachable.

hmm okay :O

 What we do is have each endpoint defined in sip.conf with
 qualify=yes and then in the dialplan use the ${SIPPEER(x)} variable
 to pull out the status of each peer and pass it into an AGI
 application to perform the load balancing etc...
 
 If you are happy with wone being a primary and one being a backup
 then if you have qualify=yes set for both you could just dial using
 the first one and then an execif hangupcause=20 then try dialing the
 backup.

okay, then i must try this way. ^^

thank you for your help and information.

greetings
dominique

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