Hi,
Can you advice SIP calls routing configuration from Kamailio
to multiple asterisk use dispatcher
I having issue if dial SIP calls landing all Asterisk server, other then
SIP calls are landing as per dispatcher
My configuration given below
*# 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();*
*}}*
----------------------------
*# -- dispatcher params for DB support --*
*modparam("dispatcher","db_url", "mysql://
openser:openserrw@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")*
*
*
With Regards
N.Prakash
On Wed, Mar 6, 2013 at 1:04 AM, Prakash N <prakas...@tevatel.com> wrote:
> 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 <shaherya...@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 <prakas...@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 <prakas...@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 <shaherya...@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 <prakas...@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 <
>>>>>> shaherya...@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 <prakas...@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 <prakas...@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@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_78...@hotmail.com
>>>>>>> Email: shaherya...@googlemail.com
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Muhammad Shahzad
>>>>> -----------------------------------
>>>>> CISCO Rich Media Communication Specialist (CRMCS)
>>>>> CISCO Certified Network Associate (CCNA)
>>>>> Cell: +49 176 99 83 10 85
>>>>> MSN: shari_78...@hotmail.com
>>>>> Email: shaherya...@googlemail.com
>>>>>
>>>>
>>>>
>>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85
>> MSN: shari_78...@hotmail.com
>> Email: shaherya...@googlemail.com
>>
>>
>
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