[asterisk-users] SIP Presence across two servers

[Leandro Dardini]

It seems very good! I am going to test it when I have a bit of time!

Leandro


2013/11/14 Ryan Wagoner <rswagoner at gmail.com>

> 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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/df35b411/attachment.html>