asterisk users - Dec 2012 - Unable to create channel of type 'SIP' (cause 20 - Unknown)

2012/12/19 Scott Huang <gyration.huang at 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 at 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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Scott Huang wrote:
> 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 at 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'
> =========================================
When you use a dynamic host type, the device needs to register to
Asterisk in order to be dialed. Otherwise there is no way to for
Asterisk to know what address to send the invite to and Asterisk will
make chan_sip issue the cause 20 error you are seeing. If the device
has a static IP and you don't want to deal with registration, you
could always change the host to that IP address. Alternatively you
could just figure out how to get your devices to register to your
Asterisk server.

--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139 

Check us out at: http://digium.com & http://asterisk.org