[asterisk-users] Multiple Asterisk

RAJNIKANT VANZA rajnivanza at gmail.com
Fri Apr 29 01:12:08 CDT 2011


Hi,

If u want to setup for 4500 or more phone then better to user OpenSER +
Asterisk.

OpenSER easily work for 10,000 calls.

You need to setup one server for OpenSER and all phone register on this
server. You need to write routing logic in OpenSER server to call connect
and if u need to play media then forward to destination asterisk server.

1 OpenSER server + Asterisk server for each location.


-- 
Best Regards,

Rajnikant Vanza
Software Engineer
-------------------------------------------------------
Working On Linux,C/C++,VoIP,Asterisk Technology


On Tue, Mar 22, 2011 at 10:47 PM, Henrique Fernandes <sf.rique at gmail.com>wrote:

> No one ?
>
> Other thing, i was reading about asterisk realtime, it can be configured to
> have multiple asterisk conectted to the same database? But how would it know
> in wich host are the "number"??
>
> Thanks!
>
> []'sf.rique
>
>
> On Wed, Mar 16, 2011 at 5:41 PM, Henrique Fernandes <sf.rique at gmail.com>wrote:
>
>> I am reading about, and some people are saying that openser is better for
>> biger envoriments, and dundi is fine for smal envoriments, does anyone have
>> any info about it ?
>>
>> We have now about 4500 convencional phones and we gonna expand a lot.
>>
>> So,
>>
>> OpenSER vs DUNDi ?
>>
>> I guess i will use Asterisk RealTime also right ?
>>
>>
>> []'sf.rique
>>
>>
>> On Wed, Mar 16, 2011 at 6:20 AM, Rizwan Hisham <rizwanhasham at gmail.com>wrote:
>>
>>> Here is a better link for DUNDi
>>>
>>>
>>> http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/
>>>
>>> skip the part which you know already
>>>
>>>
>>> On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes <sf.rique at gmail.com>wrote:
>>>
>>>>
>>>> []'sf.rique
>>>>
>>>>
>>>> On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger <pabelanger at digium.com>wrote:
>>>>
>>>>> On 11-03-15 06:19 PM, Henrique Fernandes wrote:
>>>>>
>>>>>> Have many diferenet locations that have convencional phones that need
>>>>>> to
>>>>>> call others locations with convencional phones. And we can not change
>>>>>> this,
>>>>>> I was reading and asterisk cannot handle it self this kind of setup,
>>>>>> it
>>>>>> needs an separated serrver to control and routers the calls to this
>>>>>> poins
>>>>>> right ?
>>>>>>
>>>>>> So can you guys give any help ? I guess asterisk with SER could do the
>>>>>> job ?
>>>>>>
>>>>>>  I don't believe SER will help you in the setup (see below).
>>>>>
>>>>>
>>>>>  So my question is how do i make the 2 PABX with asterisk talk to  each
>>>>>> other?  Do i need only 2 asterisk with digium or i need one server
>>>>>> with SER
>>>>>> to maki it happen ? There is another program that does what i am
>>>>>> looking for
>>>>>> ?
>>>>>>
>>>>>>  If you require local hardware for each site, then you can install
>>>>> Asterisk at each location.  You can then interconnect them using IAX2 or
>>>>> SIP, additionally you can use DUNDi in your dialplans to share information
>>>>> before the Asterisk boxes.
>>>>>
>>>>
>>>> Thanks!
>>>>
>>>> I had heard some thing about DUNDi but now i am reading i guess it is
>>>> what i need!
>>>>
>>>> I am guessing i can use both IAX2 and SIP i read something about H.323
>>>>
>>>> So i am gonna see which one is best to conect the Asterisk PBX if i am
>>>> not able to use bot SIP and IAX2
>>>>
>>>> Thanks!
>>>>
>>>> here is a link that explains better what DUNDi is!
>>>>
>>>> http://www.voip-info.org/wiki/view/DUNDi
>>>>
>>>>
>>>>> --
>>>>> Paul Belanger
>>>>> Digium, Inc. | Software Developer
>>>>> twitter: pabelanger | IRC: pabelanger (Freenode)
>>>>> Check us out at: http://digium.com & http://asterisk.org
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- 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
>>>>>
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>>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> Best Ragards
>>> Rizwan Qureshi
>>> VoIP/Asterisk Engineer
>>> Axvoice Inc.
>>>
>>> V: +92 (0) 3333 6767 26
>>> E: rizwanhasham at gmail.com
>>> W: www.axvoice.com
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>
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