On Thu, Apr 16, 2009 at 10:08 AM, Massimiliano Stucchi <[email protected]> wrote: > firmware. The problem arises when transferring a call coming in from a > SIP account to another phone. The call connects, but for the first 10 > seconds there is a situation with one-way audio, then it turns into a > fully working call. > Here are some config files: > sip.conf > > ---- > > [902] > canreinvite=no
Try enabling reinvite _______________________________________________ -- 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
