That is brought to you by the sip reinvite, in short yes, unless you set canreinvite = no to either side of that.
Apa Minerala wrote: > Am I correct in understanding that if the call comes in g729 and it is > ended in g729 ( by the provider ) , asterisk does only bridging, > therefore using very few CPU ressources ? > > Am I correct in understanding that this "bridging" means that calls ( > rtp ) pass from one provider to another, therefore using low bandwith? > > Thank you.... > > A. > > > Helping businesses save money worldwide > www.sunapemobil.ca > Numar de acces in .ro : +40318107430 (se taxeaza la pretul unui numar > 031) > Romania 3.5 c/min (USD) > Moldova 10c/min > > ------------------------------------------------------------------------ > Looking for a deal? Find great prices on flights and hotels > <http://us.rd.yahoo.com/evt=47094/*http://farechase.yahoo.com/;_ylc=X3oDMTFicDJoNDllBF9TAzk3NDA3NTg5BHBvcwMxMwRzZWMDZ3JvdXBzBHNsawNlbWFpbC1uY20-> > > with Yahoo! FareChase. > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users