Hi Krunal,
The BYE requests must be formed as a sequential request (see RFC3261)
containing the remote contact (12345 phone's contact) and a potential
Route set (RR collected on the path).
Shortly, when the GW sends the BYE, it must have in RURI either the
contact of 12345, either the RR of openser (if you did record_route()
for invite).
Regards,
Bogdan
Krunal Patel wrote:
Hi,
Here is the scenario of the call.
12345 ---> 56565656 ----> PSTN
I have registered user 12345 using sjphone.
Dialing 56565656 which is registered to the same openser.
Now i let the call ring & after timeout call is transffered to PSTN
number.
Which is terminated by PSTN gateway.
I am using mediaproxy for audio signaling & getting audio properly
both side.
Now the problem is when PSTN number hangsup the call,it should send
BYE signal to the domain name.
But instead of sending BYE signal to domain name it sends it to the IP.
& because of that the UA can't acknowledge the BYE signal & stay
connected.
& BYE keeps looping in openser.
As the pstn number sends BYE to IP , I have tried by setting
alias=xxx.xxx.xxx.xxx .
Eventhough it was not solved.
Thanks in advance
--
Regards,
Krunal Patel
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