5 Mar
2013
5 Mar
'13
5:39 p.m.
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(a)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(a)gmail.com>wrote;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(a)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(a)gmail.com>wrote;wrote:
--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
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(a)tevatel.com>wrote;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(a)tevatel.com>wrote;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_786pk(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
6 Mar
6 Mar
5:08 a.m.
New subject: Asterisk realtime with kamailio Load balancing issue for sip user
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(a)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(a)tevatel.com> wrote:
--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
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(a)gmail.com>wrote;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(a)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(a)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(a)tevatel.com>wrote;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(a)tevatel.com>wrote;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_786pk(a)hotmail.com
> Email: shaheryarkh(a)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(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
8:04 a.m.
New subject: Asterisk realtime with kamailio Load balancing issue for sip user
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(a)gmail.com>wrote;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(a)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(a)tevatel.com> wrote:
--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
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(a)gmail.com>wrote;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(a)tevatel.com>wrote;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(a)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(a)tevatel.com>wrote;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(a)tevatel.com>wrote;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_786pk(a)hotmail.com
>> Email: shaheryarkh(a)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(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
7 Mar
7 Mar
5:04 p.m.
New subject: Asterisk realtime with kamailio Load balancing issue for sip user
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 <prakash.n(a)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 <shaheryarkh(a)gmail.com>wrote;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(a)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(a)tevatel.com>wrote;wrote:
--
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk(a)hotmail.com
Email: shaheryarkh(a)googlemail.com
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(a)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(a)tevatel.com>wrote;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(a)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(a)tevatel.com>wrote;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(a)tevatel.com>wrote;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_786pk(a)hotmail.com
>>> Email: shaheryarkh(a)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(a)hotmail.com
> Email: shaheryarkh(a)googlemail.com
>
4283
days inactive
4285
days old
3 comments
2 participants
participants (2)
-
Muhammad Shahzad -
Prakash N