[OpenSER-Users] OpenSer and Asterisk-b2bua
Daniel-Constantin Mierla
daniel at voice-system.ro
Wed Jul 18 09:16:37 CEST 2007
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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>
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