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

[Nick Khamis]

Schneur,

I'm running into this problem now as well. Do you want to work on
this together?

Ninus.

On Thu, Dec 1, 2011 at 1:04 PM, Schneur Rosenberg
<rosenberg11219 at gmail.com> wrote:
> 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"
>>>
>>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users