[Asterisk-Dev] SIP codes/behaviors

[Kevin P. Fleming]

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 :-)