On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI <ad...@tootai.net> wrote: >Then create a prefix for SIP calls > >exten=>_9.,1,Dial(SIP/${EXTEN:1}) > >and you dial 9u...@domain.com from XLite > >Remember that calling sip URL is not as easy with a phone. Imagine you have an >ATA with DECT or POTS >phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the dialplan, and typed the following in XLite: 9*031...@ekiga.net This is to perform an echo test http://wiki.ekiga.org/index.php/Fun_Numbers I guess something else must be done to Asterisk for this to work: ========== CLI> -- Executing [9*031...@my-phones:1] Dial("SIP/6011-00a1b67c", "SIP/*031600") in new stack [Dec 17 11:43:14] WARNING[306]: chan_sip.c:2923 create_addr: No such host: *031600 [Dec 17 11:43:14] WARNING[306]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/6011-00a1b67c' status is 'CHANUNAVAIL' ========== I also tried this, same result: exten=>_9.,1,Dial(SIP/ippi_outgoing/${EXTEN:1}) Do I need to add something in sip.conf, or some other configuration file? Thank you. -- _____________________________________________________________________ -- 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