<snip> But then the other peer says: -- Called *31#w06123456...@xs4all-out -- SIP/xs4all-out-00000234 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/evert-00000233' status is 'CONGESTION'
Anyone an idea where i should look, or how i should change it, so that i do get a wait before sending the rest of the number to the sip peer. </snip> I don't have an answer for this but am holding my breath that *someone* does. I ran into a similar situation (dial a number, then wait, then dial an extension via SIP to PSTN) a few weeks ago and never figured out a resolution... My THOUGHT is that you would have to manually inject the DTMF into the stream somehow after the SIP provider connects the call... Thanks Dave -- _____________________________________________________________________ -- 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