[OpenSIPS-Users] Transfer problem with Opensips as a load balancer

[Schneur Rosenberg]

Bogdan there is too little info about this online, can you please help
me a bit more with this, how do I write the if statement, and how do I
set a variable for the first call, and how do I retrieve which server
was used for the first call.
On Wed, Nov 23, 2011 at 9:46 PM, Schneur Rosenberg
<rosenberg11219 at gmail.com> wrote:
> thank you Bogdan
>
> On Wed, Nov 23, 2011 at 7:32 PM, Bogdan-Andrei Iancu
> <bogdan at opensips.org> wrote:
>> Hi Schneur,
>>
>> What you have to do is to change the way you distribute the call among the
>> asterisk boxes in such a way that all calls in which a user is involved to
>> be on the same box (so that the transfers will work).
>>
>> How to do that? with a mixed routing logic. When you receive a new call, do:
>>    - check if caller or callee are already involved into an existing call on
>> a certain box. if so, route to that box
>>    - default is to do LB as you do now.
>>
>> For the check part, you need to use the dialog module (to be dialog
>> stateful), set in some dialog variables the caller / callee / box (to be
>> remembered later) and query via get_dialog_info() function -
>> http://www.opensips.org/html/docs/modules/1.7.x/dialog.html#id294051
>>
>> Regards,
>> Bogdan
>>
>> On 11/23/2011 06:48 PM, Schneur Rosenberg wrote:
>>>
>>> I'm using Opensips as a Load balancer and as a registrar, so basically
>>> all phones are registered to the Opensips, all Incoming calls hit the
>>> opensips server which forwards the call to asterisk with load
>>> balancing, asterisk decides what to do with the call ie IVR voicemail
>>> etc and if the call needs to be sent to a phone asterisk will send it
>>> back to opensips and opensips will send it to the phone.
>>>
>>> Outgoing calls are sent to asterisk via load balancing and asterisk
>>> decides how to terminate the call.
>>>
>>> This setup helps me load balance all calls and also removes the
>>> registrar load from asterisk which does not handle registrations fine
>>> when there are approx 300 peers on my asterisk system.
>>>
>>> My problem is that sometimes when I do a transfer I get back from
>>> asterisk "SIP/2.0 481 Call leg/transaction does not exist.".
>>>
>>> The test call I've done was done by calling from phone 1 a phone
>>> number which hits our system, so what happened is phone invited
>>> opensips to the DID, opensips sent the call to Asterisk server 1, then
>>> the DID called in and opensips sent it to Asterisk server 2, Asterisk
>>> server 2 saw that this did should ring on a phone so it sent it back
>>> to opensips which properly terminated the call to phone 2, then phone
>>> 1 wanted to transfer call to a outside phone, so it sent a invite to
>>> opensips with the phone number to call, opensips sent call to Asterisk
>>> server 2, then when user on phone 1 hit transfer, phone sent a refer
>>> to Asterisk 1, and asterisk 1 retuned a NOTIFY with
>>> Subscription-state: terminated;reason=noresource. and SIP/2.0 481 Call
>>> leg/transaction does not exist.
>>>
>>> Can anyone please help me solve this problem.
>>>
>>> thank you
>>> S. Rosenberg
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> OpenSIPS solutions and "know-how"
>>
>>