2012/12/19 Scott Huang <[email protected]> > Hi > > I've saw some similar case in the mail list, but seems no standard > answers, so I decide ask here again. > > Is there anyone see the message below ? I use asterisk(1.8.11-cert 9) > in my openbts2.8, and when I made a phone call, the Asterisk CLI poppd the > following messages. > > ========================================= > *CLI> == Using SIP RTP CoS mark 5 > -- Executing [8690@phones:1] Dial("SIP/IMSI466974600011287-00000000", > "SIP/IMSI466974104638690") in new stack > [Dec 18 16:00:49] WARNING[2934]: app_dial.c:2218 dial_exec_full: Unable to > create channel of type 'SIP' (cause 20 - Unknown) > == Everyone is busy/congested at this time (1:0/0/1) > -- Auto fallthrough, channel 'SIP/IMSI466974600011287-00000000' status > is 'CHANUNAVAIL' > ========================================== > > The attached files are the sip.conf and extension.conf and wireshark > trace log. > > The part of my setting in sip.conf is: > > [IMSI466974104638690] ; > callerid=8690 <8690> ; > regexten=8690 ; > canreinvite=no > type=friend > allow=gsm > context=phones > host=dynamic > registertrying=yes > > [IMSI466974102820333] ; > callerid=0333 <0333> ; > regexten=0333 ; > canreinvite=no > type=friend > allow=gsm > context=phones > host=dynamic > registertrying=yes > > > [IMSI466974600011287] ; > callerid=1287 <1287> ; > regexten=1287 ; > canreinvite=no > type=friend > allow=gsm > context=phones > host=dynamic > registertrying=yes > > The part of my setting in extensions.conf is: > > [phones] > exten => 8690,1,Dial(SIP/IMSI466974104638690) > exten => 0333,1,Dial(SIP/IMSI466974102820333) > exten => 1287,1,Dial(SIP/IMSI466974600011287) > > How to exactly configure asterisk for a sip call ? Thanks very much ! > > BR/Scott >
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