[asterisk-users] Dial SIP channel with no registration, timeout before CONGESTION?
What determines how long SIP channel waits, when you dial a peer with no registration, before returning ${DIALSTATUS} CONGESTION? When I dial a peer with no registration, SIP channel currently waits many seconds before returning ${DIALSTATUS} CONGESTION - how can I shorten this timeout? -- _ -- 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
Re: [asterisk-users] Dial SIP channel with no registration, timeout before CONGESTION?
When I dial a peer with no registration, SIP channel currently waits many seconds before returning ${DIALSTATUS} CONGESTION - how can I shorten this timeout? Look at qualify=yes for that peer. Use ChanIsAvail() before you dial. Use SIPPEER(peername|status) to check registration status. Use DB_EXISTS(SIP/Registry/peername) to check registration status. 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