[Asterisk-Users] Asterisk Dial Failover

[Douglas Garstang]

Adam,

An Audicodes Mediant 2000 gateway with a couple of PRI's. 
Why?

Doug.

-----Original Message-----
From: Adam Robins [mailto:arobins at PharmaCentra.com]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover


What are you using to terminate the PSTN calls and do the SIP
transcoding? 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover

I chose this method and have been happy with the results. 

-Jonathan

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
burke at tailorhosting.com
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Dial Failover

Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes.
One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between the boxes that goes at a configurable rate. If the Master fails
then the Slave will take over and it can even be configured for
sub-second failover. I think there is a article on voip-info.org about
this, but don't have time to look it up.

Good luck and let us know what you choose to do.

Ryan

> All,
>
> I have an Asterisk system that sends PSTN calls to an OpenSER system
to be
> routed. I have a command like this in my extensions.conf:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy,20,tr)
>
> There's actually two OpenSER systems for redundancy. I'm trying to
find a
> way to have Asterisk attempt to route the call to one OpenSER system,
and
> if it's down, fallback to another.
>
> Any first thoughts on how to achieve this?
>
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups
> are broken. If I issue a series of Dial commands, such as this:
>
> exten => 1_.,1,Dial(SIP/${EXTEN}@proxy1,20,tr)
> exten => 1_.,2,Dial(SIP/${EXTEN}@proxy2,20,tr)
>
> ... what seems to happen is that when proxy1 is down, Asterisk waits
the
> full 20 seconds before returning control. Also, This 20s includes the
time
> is takes for the other end to answer, so if I put a small value of say
5s
> in there, the dial command will probably give up before someone
answers at
> the other end. Neither is workable.
>
> Asterisk SHOULD be able to distinguish between a TRYING and no
response.
> In the event it gets no TRYING response to a dial command within a 
> specified timeout it should return control and flag an error. If on
the
> other hand it does get a TRYING response (and maybe a RINGING too) it 
> should continue to wait until the 20s has expired.
>
> I can't use dynamic DNS (ie putting two A records for a hostname in
DNS)
> because Asterisk reads the extensions.conf on startup and also seems
to
> cache what the host maps to on startup. Subsequent calls to the host 
> always return the same IP address.
>
> But... in general... how have people implemented this?
>
> Help appreciated!
> Doug
>
>
>
>
>
>
>
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