here's addition info on sip debug
11 headers, 9 lines Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format telephone-event Capabilities: us - 14, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: <sip:[EMAIL PROTECTED];ftag=as0f38e9f5;lr=on> list_route: hop: <sip:[EMAIL PROTECTED]:5028> set_destination: Parsing <sip:[EMAIL PROTECTED];ftag=as0f38e9f5;lr=on> for address/port to send to set_destination: set destination to 192.246.69.223, port 5060 sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1ecd512b4bf 00103/00000 00000ms 0000ms ULAW 192.168.1.247 2000 94915249b0e 00102/01317 00000ms 0000ms ULAW are these normal? On Sat, 2004-04-17 at 17:12, Olle E. Johansson wrote: > Chris Orme wrote: > > >>>exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r) > > Isn't the 'r' forcing a 'ringing' signal from start, regardless > of what the device you are calling are signalling. If you are calling > a SIP device, that device might return 'busy' and that's propably > why you first hear 'ringing' and then a 'busy' signal. > > I would like app_dial gurus to explain the 'r' option a bit > more so we can document it better. > > /O > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users