I some how managed to establish a call.
Callee-Opensips---Asterisk--Opensips-called.
The problem here is call is landing to opensips from Asterisk, but it is
still in loop for remaning call.
Actually, after IVR asterisk has to forward the call to opensips and
opensips has to be connect the called
(callee-opensipscalled).
Will this logic work ? How to handle it with opensips ?
I foresee following points:
1. After call is form aSterisk, oepsnips has to send BYE to asterisk and
terminate the dialog.
2. From the INVITE info from Asterisk; opensips has to re-write the
from-uri; so that instead of cal...@asterisk IP and port; contact uri should
have the one stored in location when it got registered. By this way will the
above logic work ? I dono :(
I appreciate the valuable input.
On Mon, Jul 19, 2010 at 12:21 PM, Premalatha Kuppan premala...@ngintech.com
wrote:
Hi,
I have integrated set up of opensips with TLS(1.6.2) and Asterisk (1.4.3.1)
running.
Now, my user is sending registration request via TLS, registration is
succesful; the call fails.
Here is the setup iam trying,
User (UDP) and USER with TLS gets registered with Opensips.
Asterisk performs IVR for all incoming calls to opensips.
callee--Opensips-Asterisk--opensips---called
Since Asterisk 1.4.3 doesn't support TLS; after IVR, iam tying to forward
the call to opensips.
extension.conf:
exten = s2,n,Dial(SIP/${aaa}_${bbb}_${c...@opensips:5061,20,r)
At Opensips.cfg, (Note: both opensips and Asterisk running on same box.
Opensips listening on port 5060 and 5061 and asterisk on 5070. Calls from
asterisk will always have the user uri format has aaa_bbb_ccc
if (is_method(INVITE))
{
$var(z)=$(tu{uri.user});
if($var(x)=~[0-9]+_[0-9]+_[0-9]+) {
xlog(Call from Asterisk \n);
route(1); }
}
Please help me, after IVR, i have to forward the call to opensips and
opensips should connect the callee and called.
How should i change in opensips.cfg to handle this.
I appreciate the valuable input.
Thanks,
Prem
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