i forgot to mention i use PJSIP on my asterisk........ On Thu, Feb 18, 2016 at 9:35 PM, Uri Shacked <ushac...@gmail.com> wrote:
> Hi, > > for some strange reason, ask my regulator.... i need to manipulate certain > calls. > the scenario goes like this: > > 1. caller sends invite to kamailio. > 2. kamailio transfer the call to asterisk. > 3. asterisk send progress and play "hello". > 4. asterisk creates a new call (dial) to the same kamailio with > destination callee. > 5. the callee answers the call. > > here, i need to block the 200ok. so that the caller does not receive it. > > i managed to block it with t_suspend(). > but, there is no bidirectional media. > the 183 progress was sent with sendreceive. > it seems the asterisk is waiting for the ACK in order to open both ways > for media. > > i tried to use uac_send_req() but it is being sent with no to tag. and > when i try manipulating the uac_req(turi) it does not help because it takes > all the string i entered and wraps it with <>. > > any ideas? > > BR, > Uri >
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users