Jorge, This is a function of Asterisk, not Opensips. This page may help you: http://www.voztovoice.org/?q=node/350
- Jeff On Mon, Dec 16, 2013 at 7:00 PM, Jorge Ortea <[email protected]> wrote: > Hi all, > > Suppose a platform with OpenSIPS and several Asterisk behind. A new call > in a Asterisk that send to Opensips to route to uac1. The uac1 is ringing, > it is sending 180 Ringing, then from other uac wants CallPickup this call, > this feature is dialed but when the Invite reach to OpenSIPS,,, How I can > know that Asterisk is the call? > > > Very Thanks. > Regards. > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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