<br><br><div class="gmail_quote">2012/12/19 Scott Huang <span dir="ltr"><<a href="mailto:gyration.huang@gmail.com" target="_blank">gyration.huang@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi <br><br> I've saw some similar case in the mail list, but seems no standard answers, so I decide ask here again.<br><br> 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.<br><br>=========================================<br><div>*CLI> == Using SIP RTP CoS mark 5<br>
-- Executing [8690@phones:1] Dial("SIP/IMSI466974600011287-00000000", "SIP/IMSI466974104638690") in new stack<br>[Dec 18 16:00:49] WARNING[2934]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)<br>
== Everyone is busy/congested at this time (1:0/0/1)<br> -- Auto fallthrough, channel 'SIP/IMSI466974600011287-00000000' status is 'CHANUNAVAIL'<br>==========================================<br><br> The attached files are the sip.conf and extension.conf and wireshark trace log.<br>
<br> The part of my setting in sip.conf is:<br><br><span style="color:rgb(255,0,0)">[IMSI466974104638690] ; <br>callerid=8690 <8690> ;<br>regexten=8690 ; <br>canreinvite=no<br>type=friend<br>
allow=gsm<br>
context=phones<br>host=dynamic<br>registertrying=yes<br><br>[IMSI466974102820333] ; <br>callerid=0333 <0333> ;<br>regexten=0333 ; <br>canreinvite=no<br>type=friend<br>allow=gsm<br>context=phones<br>
host=dynamic <br>registertrying=yes<br><br><br>[IMSI466974600011287] ; <br>callerid=1287 <1287> ;<br>regexten=1287 ; <br>canreinvite=no<br>type=friend<br>allow=gsm<br>context=phones<br>host=dynamic<br>
registertrying=yes</span><br><br> The part of my setting in extensions.conf is:<br><br><span style="color:rgb(255,0,0)">[phones]<br>exten => 8690,1,Dial(SIP/IMSI466974104638690)<br>exten => 0333,1,Dial(SIP/IMSI466974102820333)<br>
exten => 1287,1,Dial(SIP/IMSI466974600011287)</span><br><br> How to exactly configure asterisk for a sip call ? Thanks very much !<br><br>BR/Scott<br></div>
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