[SR-Users] Asterisk realtime with kamailio Load balancing issue for sip user

Prakash N prakash.n at tevatel.com
Wed Mar 6 07:04:34 CET 2013 

   Hi Muhammad,

      Thanks for your detail mail

       I want use Asterisk features( call Bargin, Transfer,etc ),I  am
using multiple Asterisk ,so one call comes Asterisk A box and I can able to
barge asterisk  box b  it possible only   if i  have already sent  asterisk
instances all two boxes ( Phone - Kamailio - Asterisk boxes )

As you mentioned calls are bouncing two Asterisk ,i can able understand
with your clarity mail

Can you please advice in detail  configuration  for below

1,correcting DISPATCHER and FROMASTERISK routes

2 Use asterisk instances as services bridge which are load balanced by
kamailio through dispatcher

With Regards

N.Prakash





On Wed, Mar 6, 2013 at 8:38 AM, Muhammad Shahzad <shaheryarkh at gmail.com>wrote:

> Sorry for delay, i was too busy with my work lately. Anyhow, I really
> doubt the software architecture you mentioned would scale or even work in
> the first place. Here is why,
>
> 1. You are registering same user <number-of-asterisk-instance> + 1 times,
> so if you have two asterisk behind kamailio then a single user registers on
> both asterisks as well as kamailio server. This is NOT load balancing but
> wastage of resources instead. Asterisk's capacity as SIP registrar is much
> much lower then kamailio, so whole system's capacity actually reduces down
> to asterisk capacity instead of increasing above kamailio.
>
> 2. You are using stateless forwarding, which completely disables any
> possibility of fail-over. Not only that, it will cause your calls kind
> bounce around between asterisk instances. How? its simple, user A wants to
> call user B, call comes to kamailio, which picks one asterisk instance
> through dispatcher and route calls to asterisk. When call comes to
> asterisk, it sees that user B is registered on kamailio, so it tries to
> forward call to kamailio. When call comes to kamailio, kamailio again picks
> next asterisk (due to round robin rule you are using) and send call to that
> asterisk, which again does the same thing as first asterisk, so call
> bounces between kamailio and all asterisk instance one by one till
> dispatcher list exhausts and eventually call is dropped. You may try to
> stop this by correcting DISPATCHER and FROMASTERISK routes but i guess call
> will still loop at least once.
>
> The solution is simple, forget asterisk realtime integration, use kamailio
> as registrar and proxy. Use asterisk instances as services bridge which are
> load balanced by kamailio through dispatcher.
>
> Hope this helps.
>
> Thank you.
>
>
> On Tue, Mar 5, 2013 at 4:39 PM, Prakash N <prakash.n at tevatel.com> wrote:
>
>> Hi,
>>
>>     I am facing some challenge with dispatcher configuration with two
>> Asterisk
>>
>>     I have installed Kamailio and  two Asterisk server  and Phones
>> are register with Asterisk through Kamailio
>>     I have followed this link
>> http://lists.sip-router.org/pipermail/sr-users/2011-April/068175.html
>>
>>     Now  i have added dispatcher module and dispatcher list also
>>
>>     I am try to route all calls to Asterisk with load balance
>>
>>   Can please advice the step by step configuration to route calls  from
>> Kamailio   to two Asterisk ( one call first Asterisk and Second call to
>> other asterisk )
>>
>> With Regards
>>
>> N.Prakash
>>
>>
>>
>> On Mon, Mar 4, 2013 at 10:03 AM, Prakash N <prakash.n at tevatel.com> wrote:
>>
>>>
>>> Hi Muhammad,
>>>
>>> We are following below document for Kamailio and Asterisk integration
>>>
>>>
>>> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
>>>
>>>
>>>
>>> We are plan use one Kamailio with Multiple asterisk  (Queue,IVR
>>> and Conference purpose)
>>>
>>>
>>> Now calls are landing to asterisk with load balancing using  dispatcher
>>> for Queue and IVR (One asterisk first and next Asterisk for second calls )
>>>
>>> But if try to calls extension it is landing both Asterisk server instead
>>> landing one asterisk first and next Asterisk for second calls
>>>
>>> Please advice
>>>
>>> With Regards
>>>
>>> N.Prakash
>>>
>>>
>>>
>>> On Sat, Mar 2, 2013 at 7:07 PM, Muhammad Shahzad <shaheryarkh at gmail.com>wrote:
>>>
>>>> I am not sure what you are trying to do. Your description is too brief
>>>> to understand. Can you send me complete call flow?
>>>>
>>>> Thank you.
>>>>
>>>>
>>>> On Sat, Mar 2, 2013 at 2:18 PM, Prakash N <prakash.n at tevatel.com>wrote:
>>>>
>>>>>
>>>>>    Hi Muhammad,
>>>>>
>>>>>     Thanks for your mail
>>>>>
>>>>>    Actually we are trying to do load balance with one Kamailio
>>>>> with multiple Asterisk  server
>>>>>
>>>>>    Now if call Queue,IVR to Kamailio it routing to asterisk
>>>>> with ramdam strategy load balance  ( first call on one and second to other
>>>>> server )
>>>>>   If i call extension to extension it is landing to all Asterisk ( I
>>>>> have use all Asterisk feature for that i want to route all call to asterisk
>>>>> ) on the same time ,How to do load balance  for extension calling also
>>>>>
>>>>> We are not sure what we are tiring  doi is right or wrong
>>>>>
>>>>> Please advice and correct us if anything wrong
>>>>>
>>>>> With Regards
>>>>>
>>>>> N.Prakash
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Mar 2, 2013 at 6:30 PM, Muhammad Shahzad <
>>>>> shaheryarkh at gmail.com> wrote:
>>>>>
>>>>>> Why are you forwarding instead of relaying the message to selected
>>>>>> destination? Forward is stateless and therefore likely to have NAT issues,
>>>>>> specially if destination server is behind NAT or client is behind NAT and
>>>>>> destination server is unable to handle NAT etc. etc.
>>>>>>
>>>>>> Also typically dispatcher is used to load balance calls between two
>>>>>> or more upstream server, not for load balancing extensions within one
>>>>>> server, though with some tweaking that might also be achieved but better to
>>>>>> do this kind of thing on destination server rather then on kamailio.
>>>>>>
>>>>>> Thank you.
>>>>>>
>>>>>>
>>>>>> On Sat, Mar 2, 2013 at 10:31 AM, Prakash N <prakash.n at tevatel.com>wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>>   Can you please advice for the below issue
>>>>>>>
>>>>>>> With Regards
>>>>>>>
>>>>>>> N.Prakash
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Mar 1, 2013 at 9:32 AM, Prakash N <prakash.n at tevatel.com>wrote:
>>>>>>>
>>>>>>>> Hi All,
>>>>>>>>
>>>>>>>> We have finished the Kamailio & Asterisk real time integration and
>>>>>>>> load balancing also done using dispatcher module.
>>>>>>>>
>>>>>>>> Queue and voice mails are load balancing as well.When we are
>>>>>>>> calling extension to extension it is showing in all the servers.It seems
>>>>>>>> extension are not load balancing as per our knowledge.
>>>>>>>>
>>>>>>>> I have attached the kamailio.cfg for your reference,Find
>>>>>>>> my coding below as mentioned.
>>>>>>>>
>>>>>>>> *# -- dispatcher params for DB support --*
>>>>>>>> *modparam("dispatcher","db_url", "mysql://
>>>>>>>> openser:openserrw at 192.168.1.170/openser")*
>>>>>>>> *modparam("dispatcher", "table_name", "dispatcher")*
>>>>>>>> *modparam("dispatcher", "setid_col", "setid")*
>>>>>>>> *modparam("dispatcher", "destination_col", "destination")*
>>>>>>>> *modparam("dispatcher", "flags_col", "flags")*
>>>>>>>> *modparam("dispatcher", "priority_col", "priority")*
>>>>>>>> *
>>>>>>>> *
>>>>>>>> *
>>>>>>>> -----------------------------------------------------------------------------------------
>>>>>>>> *
>>>>>>>> *# Dispatch requests*
>>>>>>>> *route[DISPATCH] {*
>>>>>>>> *if ( method=="INVITE" ) {*
>>>>>>>> *# dst_select( "GROUP", "HASH METHOD")*
>>>>>>>> *  ds_select_dst("1","4");*
>>>>>>>> *  sl_send_reply("100","Trying");*
>>>>>>>> *  forward();#uri:host, uri:port);*
>>>>>>>> *  exit();*
>>>>>>>> *}}*
>>>>>>>>
>>>>>>>> Kindly suggest the solution for the same.
>>>>>>>>
>>>>>>>> Thanks in advance.
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>>
>>>>>>>> N.Prakash
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Muhammad Shahzad
>>>>>> -----------------------------------
>>>>>> CISCO Rich Media Communication Specialist (CRMCS)
>>>>>> CISCO Certified Network Associate (CCNA)
>>>>>> Cell: +49 176 99 83 10 85
>>>>>> MSN: shari_786pk at hotmail.com
>>>>>> Email: shaheryarkh at googlemail.com
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Muhammad Shahzad
>>>> -----------------------------------
>>>> CISCO Rich Media Communication Specialist (CRMCS)
>>>> CISCO Certified Network Associate (CCNA)
>>>> Cell: +49 176 99 83 10 85
>>>> MSN: shari_786pk at hotmail.com
>>>> Email: shaheryarkh at googlemail.com
>>>>
>>>
>>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
>
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