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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
