[Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan

[Vlok Stone]

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:613 at 192.246.69.223;ftag=as0f38e9f5;lr=on>
list_route: hop: <sip:613 at 65.39.205.112:5028>
set_destination: Parsing <sip:613 at 192.246.69.223;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
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users