Hello,
use record routing (see rr module) to ensure the right path of in-dialog
requests.
Cheers.
Daniel
On 07/17/07 05:19, Ha Noi Telecommunications wrote:
Hi!
I am using OpenSer with two Asterisk-b2bua
Sip
client<--------->OpenSer<--------------------->Asterisk-b2bua<------->PSTN
|
|
<----------------------------->Asterisk-b2bua<----------->PSTN
In OpenSer configure file I am using ds_select_dst("2", "4"); to
perform load sharing the calls to PSTN.
But when Sip client hang up first, I don't konw how to make OpenSer
forward the Bye message from Sip client to correct Asterisk-b2bua to
hang up the call at PSTN side.
Can any body can help me.
Thanks and best regards
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