[asterisk-users] Get sip response code

[Robert Lister]

On Wed, May 16, 2007 at 05:46:21PM -0500, Greg Oliver wrote:
> > If I push the response code back to the handset (Cisco 7960) then it is even 
> > more unhelpful as it uses the same error message for all SIP error type 
> > response codes: "Reorder" but does not tell you why the call failed to set 
> > up. If it actually put the SIP response error on the display, that would be 
> > good, but it doesn't. (at least SIP 8.6 and prior software versions)
> 
> In order to display the response on the handset, Cisco phones require
> that you have privileges and CTI control over the phones.  The only
> un-authenticated things you can display through the phones are through
> the services and directories keys.  Unfortunately, they are keeping that
> locked up since they want you to use them with their system.  Hopefully
> they will change their minds one day.

Yes. I know that... This is exactly the limitation I am trying to work 
around, by being able to send back meaningful tones to the phone from 
asterisk in-band rather than sending back the SIP response codes which all 
get displayed by the handset as "Reorder" which is completely useless in 
informing the user what's wrong. (And the US reorder tone sounds too much 
like the UK engaged tone anyway.)

Even if the handset did display the SIP error response, I'm not expecting 
most users to understand the subtleties of what they all mean, so it seems 
better just to simplify it with a few well known tones most users are 
already familiar with (unobtainable, equipment busy, user busy, etc.) And it 
will behave in the same way regardless of the model of handset.
("Call worked/Busy/Call failed...")

Unfortunately Dial() DIALSTATUS is a bit limited in that if call setup fails 
for some reason, it mostly returns CONGESTION. Playing a congestion tone for 
perhaps 12 different call setup problems including misdials, doesn't help 
either. I want to play the right tone (for, say, unobtainable, equipment 
busy, etc.)

The ISDN gateway I am using goes to great pains to send back the correct SIP 
response to asterisk, which then just reports it as "CONGESTION" which is a 
bit limiting.

The SIP response code is displayed on asterisk's console, I just cannot see 
a way to get at it in the dial plan....

Rob