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

[Matthew Jordan]

On Thu, Aug 29, 2013 at 9:07 AM, Gareth Blades <
mailinglist+asterisk at dns99.co.uk> wrote:

> 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.
>
>
Also, a table of all of the hangup cause mappings is on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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