[Asterisk-Dev] SIP codes/behaviors

[Mark Willis]

Kevin P. Fleming wrote:

> Mark Willis wrote:
>
>> I agree. I even pointed out the paragraph in the RFC that says "zero 
>> or more 100's", but they don't care. It's "a Level3 requirement". 
>> Maybe if I go back and say others don't have to meet this requirement 
>> they will back down.
>
>
> I would try that, otherwise it would be possible to modify chan_sip to 
> _always_ send '100 Trying' if necessary.
>
>> And again, but in this case they are translating back to a PSTN code 
>> and they route advance only on certain PSTN codes. I chose 603 
>> because that mapped to a PSTN code they could handle. 503 caused them 
>> to drop the call and not try another route. I don't really care that 
>> 603 is the wrong code, I just need to enable the customer to re-route 
>> the call.
>
>
> Well, if you can _always_ use 603 instead of 503, then a simple 
> replacement in chan_sip would do it. If you need it only under certain 
> circumstances, then it would be less easy :-)


So can I just do a transmit_response(p, "100 Trying", req); right before 
the transmit_response of 404 in handle_request_invite(), or will that 
break everything? I could even add an option but I don't think this is a 
valid reason for an option.

 I already did the hack for 603 but not every customer needs that code. 
I guess it's time to code something...

Mark