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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