asterisk users - Oct 2006 - sending sip style messages in response

Hello ppl,
Is it possible to send SIP messages as response to the calling UA on 
failure, for e.g. if a number is blacklisted I would want to send 
Forbidden to the caller, not just for user comfort but also for testing 
purposes?
I see only Congestion available which sends Service Unavailable.

cheerz
- Ben

Benjamin Jacob wrote:
> Hello ppl,
> Is it possible to send SIP messages as response to the calling UA on 
> failure, for e.g. if a number is blacklisted I would want to send 
> Forbidden to the caller, not just for user comfort but also for testing 
> purposes?
> I see only Congestion available which sends Service Unavailable.
> 
Hangup(CALL_REJECTED) or Hangup(21) should work, I think.

Mikael

Nope Mikael,
* always seems to send a Decline, n also plays an unavailable file.

I tried another  scenario, A calls B, B rejects the call.
In the tcpdump I see B sending a Forbidden to *, but * sends a Service 
Unavailable to A.

hmm... not too sure, why this decision was made.


Mikael Magnusson wrote:
> Benjamin Jacob wrote:
>
>> Hello ppl,
>> Is it possible to send SIP messages as response to the calling UA on 
>> failure, for e.g. if a number is blacklisted I would want to send 
>> Forbidden to the caller, not just for user comfort but also for 
>> testing purposes?
>> I see only Congestion available which sends Service Unavailable.
>>
>
> Hangup(CALL_REJECTED) or Hangup(21) should work, I think.
>
> Mikael
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17 okt 2006 kl. 12.13 skrev Benjamin Jacob:
> Nope Mikael,
> * always seems to send a Decline, n also plays an unavailable file.
>
> I tried another  scenario, A calls B, B rejects the call.
> In the tcpdump I see B sending a Forbidden to *, but * sends a  
> Service Unavailable to A.
>
> hmm... not too sure, why this decision was made.
>
Asterisk is a multiprotocol PBX, every error that arrives to the SIP  
channel is translated
(like all other signalling) then sent to the core. The core sends it  
out again on the other
channel that translates back to SIP. In this translation, which  
follows the details specs
for ISDN to SIP translation, some granularity is lost.

/O


---
* Olle E. Johansson - oej@edvina.net
* Asterisk Training http://edvina.net/training/