[asterisk-dev] Asterisk 1.8 and SRV records
Olle E. Johansson
oej at edvina.net
Mon Apr 7 10:15:22 CDT 2014
On 07 Apr 2014, at 17:11, Eric Wieling <EWieling at nyigc.com> wrote:
> You must handle failover in the dialplan.
>
> I handle it on our systems by using an AEL script. Ugly but you are welcome to use it. See http://pastie.org/9000915
Remember that it will not work for registrations or subscriptions though...
/O
>
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Mikael Fredin
> Sent: Monday, April 07, 2014 11:03 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] Asterisk 1.8 and SRV records
>
> Thanks a lot, great information! Does this mean that I am simply out of luck regarding failover - asterisk would still try the first entry no matter if host is down or not?
>
>
>
>
> On 7 April 2014 16:37, Olle E. Johansson <oej at edvina.net> wrote:
>
>
>
> On 07 Apr 2014, at 16:09, Mikael Fredin <mikael at wiraya.com> wrote:
>
> > I have been trying to find information about this, as I found a note in the documentation that SRV records in asterisk will only work for the first entry in the record.
> >
> > All I can find is a post from Olle saying that he had it working in one of the branches - is this branch now part of the latest 1.8 version?
>
> The branch is not done yet, got postponed for some other work but will be active again soon.
> It will *never* become part of 1.8, that would be against the release regulation we have in the project.
> It is simply a very big bugfix.
>
> The current code adds "shadow peers" so we can accept calls from any server in the SRV record set.
> We also do proper selection of server on outbound calls.
>
> There's some testing of failover still to be done and an IMS hack missing.
> Read more about it here:
>
> http://svnview.digium.com/svn/asterisk/team/oej/pgtips-srv-and-outbound-stuff-1.8/README.pgtips-srv-records?revision=403237
>
> /O
>
>
> >
> > I can find nothing in the changelog regarding this.
> >
> > Would appreciate some clarity! Thank you.
> >
> > /Mikael
>
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
More information about the asterisk-dev
mailing list