[asterisk-users] How to reply with 480 Call-limit to incoming SIP call ?

[Gareth Blades]

On 29/08/13 14:42, Olivier wrote:
> Thanks for your very helpful reply.
>
> 1.My system prints out:
> CLI> core show application Hangup
>
>   -= Info about application 'Hangup' =-
>
> [Synopsis]
> Hang up the calling channel.
>
> [Description]
> This application will hang up the calling channel.
>
> [Syntax]
> Hangup([causecode])
>
> [Arguments]
> causecode
>     If a <causecode> is given the channel's hangup cause will be set
>     to the given value.
>
> [See Also]
> Answer(), Busy(), Congestion()
>
> How could we improve this Arguments section so that other Asterisk 
> admins can find available <causecode> values ?
>

Have a look in the source code in channels/chan_sip.c and you will see :-

const char *hangup_cause2sip(int cause)
{
         switch (cause) {
                 case AST_CAUSE_UNALLOCATED:             /* 1 */
                 case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: 
Can't find extension in context */
                 case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
                         return "404 Not Found";
                 case AST_CAUSE_CONGESTION:              /* 34 */
                 case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
                         return "503 Service Unavailable";
                 case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
                         return "408 Request Timeout";
                 case AST_CAUSE_NO_ANSWER:               /* 19 */
                 case AST_CAUSE_UNREGISTERED:        /* 20 */
                         return "480 Temporarily unavailable";
                 case AST_CAUSE_CALL_REJECTED:           /* 21 */
                         return "403 Forbidden";
                 case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
                         return "410 Gone";
                 case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
                         return "480 Temporarily unavailable";
                 case AST_CAUSE_INVALID_NUMBER_FORMAT:
                         return "484 Address incomplete";
                 case AST_CAUSE_USER_BUSY:
                         return "486 Busy here";
                 case AST_CAUSE_FAILURE:
                         return "500 Server internal failure";
                 case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
                         return "501 Not Implemented";
                 case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
                         return "503 Service Unavailable";
                 /* Used in chan_iax2 */
                 case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
                         return "502 Bad Gateway";
                 case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* 
Can't find codec to connect to host */
                         return "488 Not Acceptable Here";
                 case AST_CAUSE_INTERWORKING:    /* Unspecified 
Interworking issues */
                         return "500 Network error";

                 case AST_CAUSE_NOTDEFINED:
                 default:
                         ast_debug(1, "AST hangup cause %d (no match 
found in SIP)\n", cause);
                         return NULL;
         }

For any given hangup cause you can change the sip response there. For a 
list of the hangup numbers and the internal variable name look in 
include/asterisk/causes.h

So if you change chan_sip.c and add the following just before the 
'AST_CAUSE_NOTDEFINED' line and recompile and reinstall you should in 
theory be able to do a Hangup(44) to achieve what you want.

                 case AST_CAUSE_REQUESTED_CHAN_UNAVAIL:    /* 44 */
                         return "480 Temporarily Unavailable (Call limit)";

Thats only in theory. I havent tested it myself and I am not an asterisk 
developer.