Hi If I have an incoming call coming down a SIP trunk to a particular internal SIP extension- I can answer the extension fine, all works well
However, if I change extension.conf from dialling the internal extension to forward the call to an external cell phone (up the same trunk as the incoming leg of the call) I cannot get any audio and get the following error message on the console: [Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short i.e. change from [voipfone_incoming] exten => s,1,Dial(SIP/203,20,t) to [voipfone_incoming] exten => s,1,Dial(SIP/07123123...@voipfone,20,t) What's wrong?! John -- _____________________________________________________________________ -- 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
