[asterisk-users] SIP Presence across two servers
Ryan Wagoner
rswagoner at gmail.com
Thu Nov 14 06:58:33 CST 2013
I haven't tried it, but the res_corosync module states it will sync device
state across servers.
https://wiki.asterisk.org/wiki/display/AST/Corosync
On Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini <ldardini at gmail.com> wrote:
> Aligning presence over multiple servers is not simple and require some
> changes on the dialplan and some custom code to transmit the state from one
> server to the other.
>
> The BLF on the phone is displayed using the "hint" of an extension. To be
> able to manually manage the "hint" of an extension, you need to first link
> the internal hint to the Custom hint. In the extensions.conf just add:
>
> exten => _.,hint,Custom:${EXTEN}
>
> I was unable to create the same entry in the AEL language or in the
> realtime extensions table... if any was able, I will appreciate.
>
> If a phone want to know the status for the 100-TEST sip account, it will
> poll the hint for 100-TEST and in the end, it will check the status for
> Custom:100-TEST.
>
> Now you need an application to capture the change in status of every
> extension on server A and send it to server B, so the Custom:100-TEST will
> have the same value on both servers.
>
> I solved this problem creating a small pair of php application, using
> Asterisk Manager Interface to continuously listen to events. If I see a
> phone dialing out, I change its Custom state to IN_USE... if he hangups, I
> change the state back to AVAILABLE ... if it is ringing, I change the state
> in RINGING and so on. You need to take into account multiple calls can be
> made by the same phone and so it is not really so straightforward. When the
> php AMI application identify a change in the state for a phone, it notifies
> the same application running on the other server about the change, so both
> asterisk are taken aligned.
>
> Let me know if you need additional details.
>
> Leandro
>
>
>
> 2013/11/13 Lincoln King-Cliby <lincoln at controlworks.com>
>
>> Hi All,
>>
>>
>>
>> We’ve been running Asterisk for years in our offices but just recently
>> replaced an Asterisk Appliance* in our smaller office with an actual
>> server, upgraded the server in hardware in our HQ location and upgrading
>> both ends to 11.5.0 with Gareth’s patch for Cisco phones.
>>
>> 99.99% of our endpoints are Cisco 7961Gs.
>>
>>
>>
>> Each office is more-or-less standalone for ease of management and fault
>> tolerance but we have a unified dialplan and SIP “trunking” from site to
>> site via our VPN.
>>
>>
>>
>> Everything presence related works wonderfully for local users, but I’m
>> hoping there’s a way we could get presence for the people “at the other end
>> of the pipe” fairly transparently.
>>
>> We have a lot of cross-office collaboration, and our office
>> manager/receptionist (who has the battleship of a 7961G+7914+7914 BLF)
>> would love to “at a glance” know if the remote folks are available for a
>> call or not.
>>
>>
>>
>> I’m sure this has been covered, but my Googlefu us turning up a ton of
>> redundant, old, and deprecated information so I’ve resorted to asking here.
>>
>> From what I have found it sounds like it may be “easier” with IAX2 but my
>> experiments with IAX2 haven’t yielded wonderful results and management
>> prefers “SIP everywhere”
>>
>>
>>
>> If anyone has any pointers I’d greatly appreciate it – thanks in advance!
>>
>>
>>
>> Lincoln
>>
>>
>>
>> *- One of the worst IT decisions I’ve made for better or worse. Looked
>> good on paper; in practice not a good idea for anything beyond a very
>> simple SOHO.
>>
>> --
>>
>> Lincoln King-Cliby, CTS, DMC-D, CCMP-S
>>
>> Commercial Market Director
>>
>> Sr. Systems Architect | Crestron Certified Master Programmer (Silver)
>>
>> V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.com
>>
>> Crestron Services Provider
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131114/ade77edb/attachment.html>
More information about the asterisk-users
mailing list