[asterisk-users] How to quickly move on to Dahdi channels when SIP provider fails?

[klitzing at pool.informatik.rwth-aachen.de]

Hi!

> There are situations when internet connection is lost, SIP provider
> fails, or even authentication to SIP provider fails, and we want to use
> the backup Dahdi channels (PSTN). As simple as it may sound but with
> the manydifferentsituations and error messages it seems like it's not
> so easy to predict all the errors. Is there any single parameter value
> that can be changed to send the call to Dahdi instead of SIP 

There is nothing available out-of-the-box. You need to include your own IP & SIP tests in the 
dialplan before dialing out to a SIP channel. Useful for this purpose are 

- ping and host or wget, 
- GROUP() and GROUP_COUNT(), 
- SIPPEER(xxx:status), 
- CHANISAVAIL(), 
- dial timeouts and 
- post-dial error handling (see DIALSTATUS and HANGUPCAUSE as well as Asterisk 1.8 
with its ability to act directly upon the SIP response code).  

Philipp