Hi- can anyone help with this. I'm really stuck as apparently it should work. Is it a problem with the ITSP, with using the same trunk for both legs of the call etc?
John On 30 January 2010 08:57, John Taylor <[email protected]> wrote: > 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
