2012/12/19 Scott Huang gyration.hu...@gmail.com
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-,
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-' 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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