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 <[email protected]> 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 > <[email protected]> wrote: >> thank you Bogdan >> >> On Wed, Nov 23, 2011 at 7:32 PM, Bogdan-Andrei Iancu >> <[email protected]> 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 >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> OpenSIPS solutions and "know-how" >>> >>> > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
