[OpenSIPS-Users] problem with finishing call - correction
Brett Nemeroff
brett at nemeroff.com
Mon Nov 2 16:41:47 CET 2009
This normally occurs when the RURI of the BYE doesn't match the contact
header from the 200 OK. Maybe you can post those bits..
-Brett
On Fri, Oct 30, 2009 at 6:50 AM, Airton Kuada <airtonkuada at celepar.pr.gov.br
> wrote:
> Hi all, excuse me this is the correct question.
>
> Currently working with version 1.4 and testing the 1.6 to perform the
> migration, but I'm having some problems that never happened in version 1.4
> I am using to test the file opensips.cfg that comes with the package
> opensips-1.6.tar.gz, added the module "drouting" to route calls between two
> Asterisk server.
> In this configuration the system works perfectly, but when the module
> "domain" is activated (loaded and configured) the closure of a call does not
> end normally, as shown in the diagram below. If the domain module is
> disabled, everything works normally.
>
> What may be happening?
>
>
>
> *(a) opensips *(b)
> ! INVITE ! !
> !------------------>! !
>
> ! TRYING !
> !
> !<------------------! !
> ! ! INVITE !
> ! !---------------->!
> ! ! TRYING !
> ! !<----------------!
> ! ! RING !
> ! !<----------------!
> ! RING ! !
> !<------------------! !
> ! ! 200 OK !
> ! !<----------------!
> ! 200 OK ! !
> !<----------------- ! !
>
> ! ACK ! !
>
> !------------------>! !
> ! ! 200 OK !
> ! !<----------------!
> ! ! 200 OK !
> ! !<----------------!
> ! ! 200 OK !
> !
> !<----------------!
>
> ! ! (not send ack) !
> ! !================>!
>
> ! DIALOG !
> !<----------------------------------->!
>
> ! ! !
> ! BYE ! !
> !----------------->! !
>
> ! 404 Not here ! ! !
> !<--------------------! !
>
>
>
>
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>
>
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