Hi All
For the record I inspected my issue further and it was not the exact same
issue as the initial poster after all.
I am using a dispatcher for load balancing and I had nat fix up running on
both the boxes in the path of the call. This was causing add_contact_alias
to be called on both the d
Hi
I am experiencing the exact same issue.
Did you ever find a working solution?
Thank you very much for your assistance.
All the best.
Will Ferrer
Aft nix wrote
> Hi,
>
> We have the following network architecture :
>
> UAC1->kamailio>VoipSwitc
On Thu, May 24, 2012 at 6:41 PM, Vitaliy Aleksandrov
wrote:
>
> Thanks for the reply.
>
> We have already came to the same conclusion by some testing in our
> lab. It seems its a bug in provider which not constructing BYE message
> properly.
>
> But i'm interested in if its possible to detect the
Thanks for the reply.
We have already came to the same conclusion by some testing in our
lab. It seems its a bug in provider which not constructing BYE message
properly.
But i'm interested in if its possible to detect the fault in this BYE
and construct a new one and then relay it to the UAC.
On Tue, May 22, 2012 at 5:41 PM, Daniel-Constantin Mierla
wrote:
> Hello,
>
> is 205.164.40.74 the IP address of VoipSwitch? If yes, then the BYE is
> constructed to be sent back to it, because the r-uri has this IP address.
>
> R-uri in the BYE must have the IP address and port of UAC1.
>
> If yo
Hello,
is 205.164.40.74 the IP address of VoipSwitch? If yes, then the BYE is
constructed to be sent back to it, because the r-uri has this IP address.
R-uri in the BYE must have the IP address and port of UAC1.
If you give here all the sip trace for such call (from INVITE to the
BYE, taken
Hi,
We have the following network architecture :
UAC1->kamailio>VoipSwitch->PSTN-->Phone1
(Sip Client)
Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session
is terminated cleanly.
But if Phone1 hangs up the BYE message which
